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

Towards the All-IP Telephone Network at Sweden’s 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) 2

A Good Triple Play Network was Available Internet Architecture deployed by carriers to assure QoS for and control of Voice, TV and other multimedia. VoIP Mng IP-TV VoD Let’s make SIPv1 for SIP Trunking of IP-PBXs PVC3 PVC2 PVC1 PVC4 VLANs or ADSL Virtual Circuits ADSL Modem “Triple Play” LAN ATA

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

Telia SIP Connection, Business Broadband OVERVIEW Header: Relation Telia SIP Connection, Business Broadband OVERVIEW IP-PBX ADSL-modem Triple play - Bridged Telia SIP Connection Registration Signaling Telephony gateways Load balancing Port 1 Internet VoIP Port 3 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… Be DHCP client Register all accounts (all DID numbers) to Telia’s SIP platform Cut and translated from Telia presentation

There were some ISSUES IP-PBX ADSL-modem Nnn SIP Connection Add Router Header: Relation There were some ISSUES IP-PBX ADSL-modem Triple play - Bridged Nnn SIP Connection Registration Signaling Telephony gateways Load balancing Remote administration Add Router NAT? Port 1 Internet VoIP LAN WAN Port 3 If the IP-PBX can’t 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… Cut and translated from Telia presentation

Header: Relation …and a few more ISSUES IP-PBX ADSL-modem Triple play - Bridged Nnn SIP Connection Registration Signaling Telephony gateways Load balancing Remote administration Port 1 Internet Router NAT VoIP LAN WAN Port 3 More issues that have caused some headache: IP-PBX’s that only accepts calls from known servers (incompatible with load balancing) IP-PBX’s that only can register one account but expects incoming calls to all DID telephone numbers, using this single account Routers which can’t handle fragmented IP packets Remote administration over the same ADSL access Out of 10 selected PBXs, none could be used straight of! Cut and translated from Telia presentation

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

There are more things to consider… PSTN SIP Trunking Provider Network GW SIP System An E-SBC should provide: NAT/Firewall Traversal – Must NAT to same address space! Basic SIP and Network Interoperability - E.g. Authentication, Registrations, UDP/TLS/TCP, Dynamic IP address, etc. SIP Repair - E.g. Call Transfer, Fragmented packets, Bugs, etc. Features - E.g. Remote Users, Administration (remote and local) Security - LAN/PBX/VoIP network protection, Service attack protection SIP Trunk IX78 1) 2) 3) 4) 5) 2) 3) 4) 5) 2) 3) 4) 5) VoIP & Data LAN PBX Type 2 IP- PBX Few PBXs are of this type. Asterisk with firewall (IPtables /NETfilter) can be compiled and configured this way, but requires a lot. VoIP & Data LAN IP- PBX PBX Type 1 Modern IP-PBXs are of this type. Media goes directly between phone and SIP Trunk.  SIP Trunk Interface   Signaling: Media: Data LAN only PBX with system phones PBX Type 1.5

And then make it easy to install and configure 10

Confirmed Interoperability: Ingate & Intertex SIP Trunk Providers IP-PBXs 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... SIP Trunk Compliant with Carrier Equipment Acme Packet Broadsoft Genband Sonus Sylantro SER NSN More in pipeline…

The SIP Trunking Installation Wizard jkjjk

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: Telia’s 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! 13

Into the TeliaSonera Lab! Testing, integrating with management system (existing TR-069 ACS), creating a service… …and checking new PBXs PBXs

The Intertex IX78 Supports All of these Architectures! The IX78 Supports Many WAN Layer 2 and Layer 3 Architectures with QoS Separated WAN Interfaces (inherited from it’s triple play capabilities) Private Virtual Circuits E.g. Telia Internet ADSL PVC1 IP-TV VoD IMS VoIP PVC2 PVC3 E.g. Telia Internet Ethernet VLAN1 IP-TV VoD IMS VoIP VLAN2 VLAN3 Virtual LANs (VLAN) E.g. B2 Internet Ethernet WAN1 IP-TV VoD 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 The Intertex IX78 Supports All of these Architectures! 15 15

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!

IMS and More Required use of B2BUA Mode IP- PBX 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

Trunk-side Parameters (B2BUA Mode)

PBX-side Parameters (B2BUA Mode)

Registration, Call Routing, CallerID (B2BUA Mode)

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

Usage Together With Existing Firewall Also Important PSTN 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® SIP Trunk Provider GW Public Internet SIP System IP- PBX NAT/ Firewall SIParator® Data & VoIP LAN If common IP pipe, the existing firewall must restrict bandwidth usage to allow sufficient voice bandwidth. Often problematic.

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.

SIP Capable Firewalls See us at ITEXPO Room A208! Intertex Data AB www.intertex.se info@intertex.se Rissneleden 45 SE-174 44 Sundbyberg Sweden sip:reception@intertex.se Tel: +46 8 6282828 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