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Lessons Learned Across the Pond SIP Trunking Towards the All-IP Phone Network © 2010 Intertex Data AB 1 Prepared for:INTERNET TELEPHONY Conference Ingates.

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Presentation on theme: "Lessons Learned Across the Pond SIP Trunking Towards the All-IP Phone Network © 2010 Intertex Data AB 1 Prepared for:INTERNET TELEPHONY Conference Ingates."— Presentation transcript:

1 Lessons Learned Across the Pond SIP Trunking Towards the All-IP Phone Network © 2010 Intertex Data AB 1 Prepared for:INTERNET TELEPHONY Conference Ingates SIP Trunking Summit Miami, February 2011 By: Karl Erik Ståhl President & CEO Intertex Data AB Chairman Ingate Systems AB karl.stahl@intertex.se

2 © 2011 Intertex Data and Ingate Systems Confidential 2 Towards the All-IP Telephone Network at Swedens Telco, TeliaSonera The TDM network will be too expensive to maintain. ISDN (BRI) subscriber lines to be scratched first. But there are the 40-50 K (up to) 8 lines SMB PBXs (only in Sweden)

3 © 2011 Intertex Data and Ingate Systems Confidential 3 A Good Triple Play Network was Available Mng Internet PVC1 IP-TV VoD IP-TV VoD VoIP PVC2 PVC3 PVC4 LAN Architecture deployed by carriers to assure QoS for and control of Voice, TV and other multimedia. ADSL Modem Triple Play VLANs or ADSL Virtual Circuits Lets make SIPv1 for SIP Trunking of IP-PBXs ATA

4 © 2011 Intertex Data and Ingate Systems Confidential 4 Internet Telephony So Just Hook up the IP-PBX to the VoIP Pipe… TV PBX with system system phones phones SIP Trunk Interface REQUIREMENTS: As good as before (as TDM) 8 simultaneous calls, 10 – 100 numbers

5 5 Telia SIP Connection, Business BroadbandOVERVIEW Internet Telia SIP Connection Registration Signaling Telephony gateways Load balancing ADSL-modem Triple play - Bridged Port 3 Port 1 VoIP IP-PBX Internet and VoIP travel over separate channels (PVC) and are delivered on separate physical ports Different subnets for Internet and VoIP Dynamic public IP-addresses assigned to ports 1 and 3 Prioritized capacity for eight concurrent calls on the VoIP channel One DID telephone number corresponds to one SIP account SO THE PBX WITH A SIP TRUNKING INTERFACE JUST HAS TO… 1)Be DHCP client 2)Register all accounts (all DID numbers) to Telias SIP platform Cut and translated from Telia presentation

6 6 There were some ISSUES Internet Port 3 Port 1 VoIP If the IP-PBX cant act as a DHCP client, some type of NAT-router must be used between the IP-PBX and the ADSL modem. And the IP-PBX must be able to register all SIP accounts on the Nnn platform. Some method has to be used in combination with the router for SIP traversal: STUN - Simple Traversal of UDP through NATs (Network Address Translation) SIP-ALG – Application Layer Gateway But, remote administration over the same ADSL access is not possible… Add Router NAT? LAN WAN Remote administration ADSL-modem Triple play - Bridged IP-PBX Nnn SIP Connection Registration Signaling Telephony gateways Load balancing Cut and translated from Telia presentation

7 7 …and a few more ISSUES Internet Port 3 Port 1 VoIP More issues that have caused some headache: IP-PBXs that only accepts calls from known servers (incompatible with load balancing) IP-PBXs that only can register one account but expects incoming calls to all DID telephone numbers, using this single account Routers which cant handle fragmented IP packets Remote administration over the same ADSL access Router NAT LAN WAN Remote administration ADSL-modem Triple play - Bridged IP-PBX Nnn SIP Connection Registration Signaling Telephony gateways Load balancing Out of 10 selected PBXs, none could be used straight of!

8 8 Using the IX78 E-SBC solved those issues, but…

9 There are more things to consider… Data LAN only PBX with system system phones phones PBX Type 1.5 VoIP & Data LAN PBX Type 2 IP- PBX PBX Few PBXs are of this type. Asterisk with firewall (IPtables /NETfilter) can be compiled and configured this way, but requires a lot. An E-SBC should provide: 1)NAT/Firewall Traversal – Must NAT to same address space! 2)Basic SIP and Network Interoperability - E.g. Authentication, Registrations, UDP/TLS/TCP, Dynamic IP address, etc. 3)SIP Repair - E.g. Call Transfer, Fragmented packets, Bugs, etc. 4)Features - E.g. Remote Users, Administration (remote and local) 5)Security - LAN/PBX/VoIP network protection, Service attack protection VoIP & Data LAN IP- PBX PBX PBX Type 1 Modern IP-PBXs are of this type. Media goes directly between phone and SIP Trunk. SIP Trunk Interface Signaling: Media: SIP Trunk PSTN SIP Trunking Provider Network GW SIP System 2) 3) 4) 5) IX781) 2) 3) 4) 5)

10 10 And then make it easy to install and configure

11 © 2011 Intertex Data and Ingate Systems Confidential Confirmed Interoperability: Ingate & Intertex SIP Trunk Providers IP-PBXs SIP Trunk Compliant with Aastra Aastra/Ericsson MX One Adtran UC Server Digium/Asterisk Avaya Aura Avaya IP Office Avaya SES/CM Avaya QE Brekeke Broadsoft Cisco Fonality HP/3Com -VCX Innovaphone Interactive Intelligence Iwatsu LG Nortel Microsoft OCS Mitel NEC / Sphere Nortel BCM Nortel SCS Objectworld Panasonic Samsung SER Shoretel Siemens SIP-Gear Swyx More in pipeline.... 360 Networks Airespring AT&T BandTel Bandwidth.com Broadvox BT (British Telecom) Cablevision Cbeyond Cellip Comm Partners Cordia Corporation Deltacom Excel Switching Gamma Telecom GEOS Global Crossing IP-Only Nectar Level 3 Netlogic Netsolutions Nexvortex Nuvox O1 One Communications Paetec Primus RNK Telecom Skype TDC Telavox Tele2 Tele Pacific Teletek TeliaSonera Toplink Tritel VoEX Voice Flex VoIP Unlimited Voxbone Voxitas XeloQ More in pipeline... Carrier Equipment Acme Packet Broadsoft Genband Sonus Sylantro SER NSN More in pipeline…

12 © 2011 Intertex Data and Ingate Systems Confidential The SIP Trunking Installation Wizard jkjjk

13 © 2011 Intertex Data and Ingate Systems Confidential 13 All worked fine – But, time to make SIPv2! New SIP IMS platform Will take over generally for the future Higher scale But more complex SIP interface More IP delivery networks ADSL2+ AnnexM: Triple play as before FiberLAN: 100 Mbps Ethernet triple play (VLAN tagged) Prolane: Internet with priority VoIP channel Internet: Telias SIP Trunking over other providers Internet access Up to 60 simultaneous calls per trunk group (8 in SIPv1) CPE / E-SBC comes with the service, owned by Telia Provisioning and management by Telia Reused ACS (TR-069 management system) for residential Combined with Intertex PBX selection Wizard What is required from the CPE / E-SBC? Intertex IX78 still the choice!

14 © 2011 Intertex Data and Ingate Systems Confidential Into the TeliaSonera Lab! Testing, integrating with management system (existing TR-069 ACS), creating a service… …and checking new PBXs PBXs

15 © 2011 Intertex Data and Ingate Systems Confidential 15 The IX78 Supports Many WAN Layer 2 and Layer 3 Architectures with QoS Separated WAN Interfaces (inherited from its triple play capabilities) The Intertex IX78 Supports All of these Architectures! Private Virtual Circuits E.g. Telia Internet ADSL PVC1 IP-TV VoD IP-TV VoD IMS VoIP IMS VoIP PVC2 PVC3 E.g. Telia Internet Ethernet VLAN1 IP-TV VoD IP-TV VoD IMS VoIP IMS VoIP VLAN2 VLAN3 Virtual LANs (VLAN) E.g. B2 Internet Ethernet WAN1 IP-TV VoD IP-TV VoD IMS VoIP IMS VoIP WAN2 WAN3 IP QoS Separated Subnets IP Level QoS E.g. BT Internet ADSL or Ethernet Priority3 Priority2 Priority1 IMS VoIP IP-TV VoD

16 © 2011 Intertex Data and Ingate Systems Confidential 16 Performance and Call Handling Capacity Over 50 simultaneous calls (20 ms voice packets) carrying media Call rate of 8 calls/s in proxy mode and 3 calls/s in B2BUA mode. (more than required to support 50 simultaneous calls) Up to 255 registrations. SIP end-points can be more. CPU Usage: 60 simultaneous calls without MOS degeneration were reached!

17 © 2011 Intertex Data and Ingate Systems Confidential 17 IMS and More Required use of B2BUA Mode Proxy Mode IP-PBX talks to Service Registration/Authentication model must match Little configuration in the IX78 Service credentials in the PBX B2BUA Mode (Proxy still doing the basics) IP-PBX only talks to the IX78 Wider separation between PBX and Service Service Credentials only in the IX78 More SIP Normalization possibilities (e.g. REFER) Any new operator service platform only requires IX78 reconfiguration (the PBX configuration can remain) IP- PBX IP- PBX

18 © 2011 Intertex Data and Ingate Systems Confidential 18 Trunk-side Parameters (B2BUA Mode)

19 © 2011 Intertex Data and Ingate Systems Confidential 19 PBX-side Parameters (B2BUA Mode)

20 © 2011 Intertex Data and Ingate Systems Confidential 20 Registration, Call Routing, CallerID (B2BUA Mode)

21 © 2011 Intertex Data and Ingate Systems Confidential 21 PSTN Public Internet SIP Trunking Provider GW SIP System Data & VoIP LAN IP-PBX Demarcation point of service and bringing SIP communication to the LAN Soft Clients and Multimedia Terminals Intertex IX78 Remote Users Support for UC LAN and Multimedia Terminals as well as Remote Users

22 © 2011 Intertex Data and Ingate Systems Confidential Usage Together With Existing Firewall Also Important PSTN Public Internet SIP Trunk Provider GW SIP System IP- PBX NAT/ Firewall Data & VoIP LAN If common IP pipe, the existing firewall must restrict bandwidth usage to allow sufficient voice bandwidth. Often problematic. PSTN Public Internet SIP Trunk Provider GW SIP System IP- PBX NAT/ Firewall Bridge for Existing NAT/ Firewall (non SIP aware) Data & VoIP LAN WAN SIParator mode allows the Ingate or Intertex to control data usage on the Pipe to assure sufficient voice bandwidth! WAN SIParator® SIParator® 22

23 © 2011 Intertex Data and Ingate Systems Confidential Jonas Östergren, TeliaSonera, Interviewed Live from Sweden Using Omnitor application Allan eC: Voice: G.722 wide band codec Video: H.264 300kbps Real-time text: RFC4103 Using standard SIP over the Internet.

24 © 2011 Intertex Data and Ingate Systems Confidential 24 SIP Capable Firewalls Ingate Systems Inc. www.ingate.com Info@ingate.com 7 Farley Road Hollis NH 03049 United States Ph: +1 (603) 883-6569 Ph Sweden: +46 8 6007750 Intertex Data AB www.intertex.se info@intertex.se Rissneleden 45 SE-174 44 Sundbyberg Sweden sip:reception@intertex.se Tel: +46 8 6282828 See us at ITEXPO Room A208!


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