User manual ALCATEL-LUCENT OMNIACCES - VOIP GATEWAY

Part No. 060170-10, Rev. C April 2004 VoIP User Manual Release 4.5 An Alcatel service agreement brings your company the assurance of 7x24 no-excuses technical support. You'll also receive regular software updates to maintain and maximize your Alcatel product's features and functionality and on-site hardware replacement through our global network of highly qualified service delivery partners. Additionally, with 24-hour-a-day access to Alcatel's Service and Support web page, you'll be able to view and update any case (open or closed) that you have reported to Alcatel's technical support, open a new case or access helpful release notes, technical bulletins, and manuals. For more information on Alcatel's Service Programs, see our web page at www.ind.alcatel.com, call us at 1-800-995-2696, or email us at support@ind.alcatel.com. This manual documents Release 4.5 Voice over IP (VoIP) hardware and software. The functionality described in this manual is subject to change without notice. Copyright© 2004 by Alcatel Internetworking, Inc. All rights reserved. This document may not be reproduced in whole or in part without the express written permission of Alcatel Internetworking, Inc. Alcatel® and the Alcatel logo are registered trademarks of Alcatel. Xylan®, OmniSwitch®, PizzaSwitch® and OmniStack® are registered trademarks of Alcatel Internetworking, Inc. AutoTrackerTM, OmniAccessTM, OmniCoreTM, Omni Switch/RouterTM, OmniVistaTM, PizzaPortTM, PolicyViewTM, RouterViewTM, SwitchManagerTM, SwitchStartTM, VoiceViewTM, WANViewTM, WebViewTM, X-CellTM, X-VisionTM and the Xylan logo are trademarks of Alcatel Internetworking, Inc. All-In-OneSM is a service mark of Alcatel Internetworking, Inc. All other brand and product names are trademarks of their respective companies. 26801 West Agoura Road Calabasas, CA 91301 (818) 880-3500 FAX (818) 880-3505 info@ind.alcatel.com US Customer Support­(800) 995-2696 International Customer Support­(818) 878-4507 Internet­http://eservice.ind.alcatel.com Cautions FCC Compliance: This equipment has been tested and found to comply with the limits for Class A digital device pursuant to Part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a commercial environment. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instructions in this guide, may cause interference to radio communications. Operation of this equipment in a residential area is likely to cause interference, in which case the user will be required to correct the interference at his own expense. The user is cautioned that changes and modifications made to the equipment without approval of the manufacturer could void the user's authority to operate this equipment. It is suggested that the user use only shielded and grounded cables to ensure compliance with FCC Rules. This equipment does not exceed Class A limits per radio noise emissions for digital apparatus, set out in the Radio Interference Regulation of the Canadian Department of Communications. Avis de conformité aux normes du ministére des Communications du Canada Cet équipement ne dépasse pas les limites de Classe A d'émission de bruits radioélectriques pour les appareils numériques, telles que prescrites par le Réglement sur le brouillage radioélectrique établi par le ministére des Communications du Canada. Lithium Batteries Caution: There is a danger of explosion if the Lithium battery in your chassis is incorrectly replaced. Replace the battery only with the same or equivalent type of battery recommended by the manufacturer. Dispose of used batteries according to the manufacturer's instructions. The manufacturer's instructions are as follows: Return the module with the Lithium battery to Alcatel. The Lithium battery will be replaced at Alcatel's factory. page iii page iv Table of Contents 1 VoIP Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-1 VoIP Networks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-1 Getting Started with VoIP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-2 VoIP Telephone Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-4 A VoIP Call Scenario . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-4 Elements of a Converged Network . . . . . . VoIP H.323 Client . . . . . . . . . . . . VoIP H.323 Gateway . . . . . . . . . . VoIP H.323 Gatekeeper (Optional) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-7 1-7 1-7 1-8 H.323 VoIP Gateway Voice and Convergence Features . . . . . . . . . . . . . . . . . Signaling Control and Voice Interoperability (Voice Features) . . . . . . . . . H.323 Call Control and Network Interoperability (Convergence Features) H.323 Network Call Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Alcatel VoIP Network Dialing Schemes (AVNDS) . . . . . . . . . . . . . . . Switch Backplane Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-10 1-11 1-15 1-15 1-15 1-16 VoIP Standards for Development . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-17 Codec Support (G.711, G.723.1, G.729a) . . . . . . . . . . . . . . . . . . . . . . . . . 1-17 VON (Voice on the Net) Developments . . . . . . . . . . . . . . . . . . . . . . . . . 1-18 VoIP and VLANs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-19 2 VoIP Daughtercards . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-1 VoIP Daughtercard Types . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-2 Digital Signal Processors (DSPs), DIMMs and Available Channels . . . . . . . . 2-4 Voice Switching Daughtercard -- Digital VSD Front Panel . . . . . . . . . . . . VSD Deadman Switch . . . . . . . . VSD Cross-Over Toggle Switch . . VSD Pinouts . . . . . . . . . . . . . . . VSD Jumpers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-6 . 2-7 . 2-8 . 2-9 2-11 2-12 2-13 2-13 2-13 2-13 2-13 2-15 2-16 Voice Switching Daughtercard -- Euro BRI ISDN . . . . . . . . . . . Digital Signal Processors (DSPs) and Available Channels VSB Deadman Switch . . . . . . . . . . . . . . . . . . . . . . . . . VSB NT (LT)/TE Cross-Over Toggle Switch . . . . . . . . . . VSB Pinouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . VSB Front Panel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . VSB Jumpers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . page v Table of Contents Voice Switching Daughtercard -- Analog . . . . . . . . . . . . . . . . . . . . . . . . VSA Front Panel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . VSA Pinouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . VSAs and Digital Signal Processors (DSPs), DIMMs and Available Channels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . VSAs and the Deadman Switch . . . . . . . . . . . . . . . . . . . . . . . . . . VSAs and Cross-Over Toggle Switches . . . . . . . . . . . . . . . . . . . . VSA Jumpers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-19 . . . . . . 2-21 . . . . . . 2-22 . . . . . . . . . . . . . . . . . . . . . . . . 2-22 2-22 2-22 2-23 VSX Switching Module . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-25 VSX Technical Specifications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-25 VoIP Daughtercard Port Numbering Schemes . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-28 3 Network Dialing Schemes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-1 How to Select a Network Dialing Scheme (AVNDS) . . . . . . . . . . . . . . . . . . . . 3-3 Network Dialing Scheme VoIP Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-5 VoIP Networks without PSTN -- Example 1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-7 Four Digit Extensions and Two Voice Daughtercards . . . . . . . . . . . . . . . . . . . 3-7 VoIP Networks without PSTN -- Example 2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-9 Trunk Groups and Three Voice Daughtercards . . . . . . . . . . . . . . . . . . . . . . . . 3-9 VoIP Networks without PSTN -- Example 3 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-11 One Hunt Group (48 Channels Across Two T1s) . . . . . . . . . . . . . . . . . . . . . 3-11 VoIP Networks without PSTN -- Example 4 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-13 One Hunt Group (60 Channels Across Two E1s) . . . . . . . . . . . . . . . . . . . . . 3-13 VoIP Networks without PSTN -- Example 5 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-15 One Hunt Group (96 Channels Across Four T1s) . . . . . . . . . . . . . . . . . . . . . 3-15 VoIP Networks without PSTN -- Example 6 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-17 One Hunt Group (144 Channels Across Six T1s) . . . . . . . . . . . . . . . . . . . . . . 3-17 VoIP Networks without PSTN -- Example 7 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-19 Four Hunt Groups (12 Channels Per Hunt Group) . . . . . . . . . . . . . . . . . . . . 3-19 VoIP Networks without PSTN -- Example 8 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-21 48 Individual Hunt Groups (One Channel Per Group) . . . . . . . . . . . . . . . . . 3-21 VoIP Networks without PSTN -- Example 9 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-23 Trunk Groups and Mixed Length Extensions . . . . . . . . . . . . . . . . . . . . . . . . 3-23 VoIP Networks without PSTN -- Example 10 . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-25 Strip Digit Length (2) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-25 VoIP Networks without PSTN -- Example 11 . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-27 Trunk Groups and Eleven Digit Extensions . . . . . . . . . . . . . . . . . . . . . . . . . 3-27 VoIP Networks without PSTN -- Example 12 . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-29 H.323 Gatekeeper . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-29 VoIP Networks with PSTN -- Example 13 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-31 North American PSTN and VoIP Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-31 page vi Table of Contents VoIP Networks with PSTN -- Example 14 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-37 North American PSTN, International PSTN and VoIP Calls . . . . . . . . . . . . . . . 3-37 VoIP Networks with PSTN -- Example 15 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-43 PSTN and Eleven Digit Extensions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-43 VoIP Networks with PSTN -- Example 16 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-45 FAX over IP Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-45 VoIP Networks with PSTN -- Example 17 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-47 Mixed Digital and Analog Voice Daughtercards . . . . . . . . . . . . . . . . . . . . . . 3-47 VoIP Networks with PSTN -- Example 18 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-49 Caller ID (Static) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-49 VoIP Networks with Interoperability -- Example 19 . . . . . . . . . . . . . . . . . . . . . . 3-51 H.323 Gateway to Microsoft NetMeeting (without FastStart) . . . . . . . . . . . . . . 3-51 VoIP Networks with Interoperability -- Example 20 . . . . . . . . . . . . . . . . . . . . . . 3-53 H.323 Gateway to Cisco Router . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-53 VoIP Networks with Interoperability -- Example 21 . . . . . . . . . . . . . . . . . . . . . . 3-55 H.323 Gateway to OmniPCX 4400 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-55 VoIP Networks with Interoperability -- Example 22 . . . . . . . . . . . . . . . . . . . . . . 3-57 OmniPCX 4400 and E1 QSIG . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-57 VoIP Networks with Interoperability -- Example 23 . . . . . . . . . . . . . . . . . . . . . . 3-59 OmniPCX and Euro PRI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-59 VoIP Networks with Interoperability -- Example 24 . . . . . . . . . . . . . . . . . . . . . . 3-61 Other PBXs with T1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-61 VoIP Networks with Interoperability -- Example 25 . . . . . . . . . . . . . . . . . . . . . . 3-63 Other PBXs with Euro BRI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-63 VoIP Networks with Interoperability -- Example 26 . . . . . . . . . . . . . . . . . . . . . . 3-65 Mixed European Digital and Analog Voice Daughtercards . . . . . . . . . . . . . . . 3-65 AVNDS Master List of Features by CLI Command . . . . . . . . . . . . . . . . . . . . . . . . 3-67 4 Setup and Installation ................ Components of VoIP . . . . . . . . . . . . . . . . . Assumptions and Recommendations . . . . . Configuration Restrictions . . . . . . . . . . . . . General Installation Procedures . . . . . . . . . Instructions for Additional VoIP Installations Example VSM Boot File (vsmboot.asc) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-1 4-1 4-2 4-3 4-4 4-7 4-8 page vii Table of Contents 5 VoIP Commands . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-14 5-16 5-17 5-18 5-19 5-20 5-23 5-24 5-25 5-26 5-27 5-28 5-29 5-30 5-31 5-32 5-33 5-34 5-36 5-38 5-39 5-40 5-41 5-42 5-43 5-44 5-45 5-46 5-47 5-48 5-49 5-50 5-51 5-53 5-54 5-55 5-57 5-58 5-59 5-60 5-61 Voice Switching Daughtercard Commands . . . . voice daughter card ip mask . . . . . . . . . . . voice daughter card ip address . . . . . . . . . voice daughter card ip default gateway . . . voice daughter card activate . . . . . . . . . . . voice dump . . . . . . . . . . . . . . . . . . . . . . . . voice daughter card h.323 out fast start . . . voice daughter card h.323 in fast start . . . . voice daughter card h.323 auto answer . . . voice daughter card first digit wait duration voice daughter card inter digit wait duration voice daughter card dial time wait duration voice daughter card termination digit . . . . . voice daughter card cadence coefficient . . . voice daughter card ring id . . . . . . . . . . . . voice daughter card vsb clock source . . . . . voice daughter card vsb external clock port voice port interface type . . . . . . . . . . . . . . voice port frame format . . . . . . . . . . . . . . . voice port circuit identifier . . . . . . . . . . . . . voice port nfas framing . . . . . . . . . . . . . . . voice port line build out . . . . . . . . . . . . . . voice port line length . . . . . . . . . . . . . . . . voice port attenuation . . . . . . . . . . . . . . . . voice port cable type . . . . . . . . . . . . . . . . . voice port line coding . . . . . . . . . . . . . . . . voice port facilities data link protocol . . . . . voice port facilities data link port role . . . . voice port transmit clock source . . . . . . . . . voice port loop back mode . . . . . . . . . . . . voice port signaling mode . . . . . . . . . . . . . voice port trap generation . . . . . . . . . . . . . voice port isdn protocol . . . . . . . . . . . . . . voice port isdn switch type . . . . . . . . . . . . voice port bri line type . . . . . . . . . . . . . . . voice channel isdn d channel . . . . . . . . . . . voice channel isdn b channel . . . . . . . . . . . Channel Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . voice channel mode . . . . . . . . . . . . . . . . . . . . . . . . voice channel dial in private line automatic ringdown voice channel state . . . . . . . . . . . . . . . . . . . . . . . . . page viii Table of Contents Telephony Signaling Attributes . . . . . . . . . . . . . . . . view voice signaling channel . . . . . . . . . . . . . . . voice signaling protocol . . . . . . . . . . . . . . . . . . . voice signaling out wait . . . . . . . . . . . . . . . . . . . voice signaling out tone digit duration . . . . . . . . voice signaling out tone interdigit duration . . . . . voice signaling out dialing type . . . . . . . . . . . . . voice signaling call duration limit . . . . . . . . . . . . voice signaling answer wait limit . . . . . . . . . . . . voice signaling hang up wait limit . . . . . . . . . . . voice signaling fax holdover . . . . . . . . . . . . . . . voice signaling companding . . . . . . . . . . . . . . . . voice signaling receive gain . . . . . . . . . . . . . . . . voice signaling transmit gain . . . . . . . . . . . . . . . voice signaling idle noise . . . . . . . . . . . . . . . . . . voice signaling em on hook debounce . . . . . . . . voice signaling em off hook debounce . . . . . . . . voice signaling em seize detect . . . . . . . . . . . . . voice signaling em clear detect . . . . . . . . . . . . . voice signaling em clear confirm detect . . . . . . . voice signaling em clear confirm wait max . . . . . voice signaling em guard all . . . . . . . . . . . . . . . voice signaling em guard out . . . . . . . . . . . . . . . voice signaling em dial tone . . . . . . . . . . . . . . . voice signaling em min connection time . . . . . . . voice signaling em hang up wait . . . . . . . . . . . . voice signaling emw in wink wait min . . . . . . . . voice signaling emw in wink wait max . . . . . . . . voice signaling emw in wink duration . . . . . . . . voice signaling emw in wink digit ignore . . . . . . voice signaling emw out wink wait max . . . . . . . voice signaling emw out wink duration min . . . . voice signaling emw out wink duration max . . . voice signaling emi glare report . . . . . . . . . . . . . voice signaling emi digit wait . . . . . . . . . . . . . . . voice signaling emd in delay min . . . . . . . . . . . . voice signaling emd in delay max . . . . . . . . . . . voice signaling emd in digit ignore . . . . . . . . . . voice signaling emd out integrity check . . . . . . . voice signaling emd out delay duration min . . . . voice signaling emd out detail duration max . . . . voice signaling emd out delay check . . . . . . . . . voice signaling fxs ls on hook debounce . . . . . . voice signaling fxs ls off hook debounce . . . . . . voice signaling fxs ls seize detect . . . . . . . . . . . . voice signaling fxs ls originate clear detect . . . . . voice signaling fxs ls answer clear detect . . . . . . voice signaling fxs ls supervisory disconnect wait . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-62 . 5-67 . 5-68 . 5-69 . 5-70 . 5-71 . 5-72 . 5-73 . 5-74 . 5-75 . 5-76 . 5-77 . 5-78 . 5-79 . 5-80 . 5-81 . 5-82 . 5-83 . 5-84 . 5-85 . 5-86 . 5-87 . 5-88 . 5-89 . 5-90 . 5-91 . 5-92 . 5-93 . 5-94 . 5-95 . 5-96 . 5-97 . 5-98 . 5-99 .5-100 .5-101 .5-102 .5-103 .5-104 .5-105 .5-106 .5-107 .5-108 .5-109 .5-110 .5-111 .5-112 .5-113 page ix Table of Contents voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice voice signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling fxs ls supervisory disconnect duration . . . . . . . fxs ls caller id . . . . . . . . . . . . . . . . . . . . . . . . fxs ls ringing debounce . . . . . . . . . . . . . . . . . fxo ls supervisory disconnect detection . . . . . . fxo ls supervisory disconnect . . . . . . . . . . . . . fxo ls guard out . . . . . . . . . . . . . . . . . . . . . . . fxo ls ringing inter cycle . . . . . . . . . . . . . . . . . fxo ls ringing inter pulse . . . . . . . . . . . . . . . . . fxo ls caller id . . . . . . . . . . . . . . . . . . . . . . . . fxo ls answer after . . . . . . . . . . . . . . . . . . . . . fxo ls loop current debounce . . . . . . . . . . . . . fxo ls battery reversal debounce . . . . . . . . . . . fxs gs seize detect . . . . . . . . . . . . . . . . . . . . . fxs gs on hook debounce . . . . . . . . . . . . . . . . fxs gs originate clear detect . . . . . . . . . . . . . . fxs gs answer clear detect . . . . . . . . . . . . . . . . fxs gs min ring ground . . . . . . . . . . . . . . . . . . fxs gs max wait loop . . . . . . . . . . . . . . . . . . . fxs gs min loop open . . . . . . . . . . . . . . . . . . . fxs gs caller id . . . . . . . . . . . . . . . . . . . . . . . . fxs gs off hook debounce . . . . . . . . . . . . . . . . fxs gs ring ground debounce . . . . . . . . . . . . . fxs gs ring id . . . . . . . . . . . . . . . . . . . . . . . . . fxo gs connection loop open debounce . . . . . fxo gs max tip ground wait . . . . . . . . . . . . . . . fxo gs tip ground debounce . . . . . . . . . . . . . . fxo gs ringing debounce . . . . . . . . . . . . . . . . . fxo gs ringing inter cycle . . . . . . . . . . . . . . . . fxo gs ringing inter pulse . . . . . . . . . . . . . . . . fxo gs caller id detection . . . . . . . . . . . . . . . . fxo gs answer after . . . . . . . . . . . . . . . . . . . . . fxo gs loop current debounce . . . . . . . . . . . . . fxo gs battery reversal debounce . . . . . . . . . . caller id name . . . . . . . . . . . . . . . . . . . . . . . . caller id number . . . . . . . . . . . . . . . . . . . . . . . tone table . . . . . . . . . . . . . . . . . . . . . . . . . . . call progress tone . . . . . . . . . . . . . . . . . . . . . . call progress tone detection configuration . . . . v.18 tone detection threshold hang time . . . . . v.18 tone detection threshold level . . . . . . . . . v.18 tone detection threshold fraction . . . . . . . single frequency tone detection threshold level single frequency tone detection threshold time echo canceller non-linear sensitivity . . . . . . . . acoustic echo canceller mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-114 .5-115 .5-116 .5-117 .5-118 .5-119 .5-120 .5-121 .5-122 .5-123 .5-124 .5-125 .5-126 .5-127 .5-128 .5-129 .5-130 .5-131 .5-132 .5-133 .5-134 .5-135 .5-136 .5-137 .5-138 .5-139 .5-140 .5-141 .5-142 .5-143 .5-144 .5-145 .5-146 .5-147 .5-148 .5-149 .5-150 .5-151 .5-152 .5-153 .5-154 .5-155 .5-156 .5-157 .5-158 page x Table of Contents voice voice voice voice voice voice voice voice voice voice voice voice voice signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling signaling acoustic echo canceller non-linear processor . . . acoustic echo canceller output . . . . . . . . . . . . . acoustic echo canceller handset speaker gain . . acoustic echo canceller hands free speaker gain override in band call progress tones . . . . . . . . . override full call progress tones . . . . . . . . . . . . override ring back . . . . . . . . . . . . . . . . . . . . . . override in band codec switching . . . . . . . . . . . override psu codec switching . . . . . . . . . . . . . . override network overlap dialing . . . . . . . . . . . override information element transport . . . . . . . override qsig information element transport . . . override setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-159 .5-160 .5-161 .5-162 .5-163 .5-164 .5-165 .5-166 .5-167 .5-168 .5-169 .5-170 .5-171 .5-172 .5-174 .5-175 .5-176 .5-180 .5-181 .5-182 .5-183 .5-185 .5-186 .5-187 .5-188 .5-189 .5-190 .5-191 .5-192 .5-193 .5-194 .5-195 .5-196 .5-197 .5-198 .5-199 .5-200 .5-201 .5-202 .5-203 .5-204 .5-205 .5-206 .5-207 .5-208 .5-209 Coding Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . voice coding profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . voice no coding profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . view voice coding profile . . . . . . . . . . . . . . . . . . . . . . . . . . . voice coding profile all reset . . . . . . . . . . . . . . . . . . . . . . . . . voice channel available coding profile . . . . . . . . . . . . . . . . . . voice channel assign preferred coding profile . . . . . . . . . . . . . voice coding profile coding type . . . . . . . . . . . . . . . . . . . . . . voice coding profile voice packet interval . . . . . . . . . . . . . . . . voice coding profile voice network delay buffer mode . . . . . . voice coding profile voice network delay buffer nominal delay voice coding profile voice network delay buffer max delay . . . voice coding profile voice activity detector . . . . . . . . . . . . . . . voice coding profile voice activity detection threshold mode . . voice coding profile voice activity detection threshold level . . voice coding profile voice dtmf relay . . . . . . . . . . . . . . . . . . . voice coding profile switchover . . . . . . . . . . . . . . . . . . . . . . . voice coding profile call progress tone detection . . . . . . . . . . voice coding profile voice dtmf relay . . . . . . . . . . . . . . . . . . . voice coding profile single frequency tone detection . . . . . . . voice coding profile voice echo canceller . . . . . . . . . . . . . . . . voice coding profile voice echo canceller non linear . . . . . . . . voice coding profile voice echo canceller comfort noise mode voice coding profile echo canceller noise level . . . . . . . . . . . . voice coding profile voice echo canceller tail length . . . . . . . . voice coding profile echo canceller refresh configuration . . . . voice coding profile echo canceller refresh state . . . . . . . . . . . voice coding profile fax rate . . . . . . . . . . . . . . . . . . . . . . . . . voice coding profile fax transmit level . . . . . . . . . . . . . . . . . . voice coding profile fax carrier detect threshold . . . . . . . . . . . voice coding profile fax timeout . . . . . . . . . . . . . . . . . . . . . . . voice coding profile fax t.38 high speed packet rate . . . . . . . . voice coding profile fax t.38 low speed redundancy . . . . . . . . page xi Table of Contents voice voice voice voice voice voice coding coding coding coding coding coding profile profile profile profile profile profile fax t.38 high speed redundancy . . . fax t.38 training check field method silence detect time . . . . . . . . . . . . . silence detect level . . . . . . . . . . . . . g.711 modem resampling mode . . . caller id . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-210 .5-211 .5-212 .5-213 .5-214 .5-215 .5-216 .5-217 .5-218 .5-219 .5-220 .5-221 .5-222 .5-223 .5-224 .5-225 .5-226 .5-228 .5-230 .5-231 .5-232 .5-233 .5-235 .5-236 .5-237 .5-238 .5-239 .5-240 .5-243 .5-244 .5-245 .5-246 .5-247 .5-248 .5-249 .5-250 .5-251 .5-253 .5-255 .5-256 .5-257 .5-258 Voice Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . voice network card h.323 gatekeeper control . . . . . . . . . . . . voice network h.323 gatekeeper mode . . . . . . . . . . . . . . . . . voice network h.323 gatekeeper address . . . . . . . . . . . . . . . voice network h.323 allow calls without gatekeeper . . . . . . . voice network h.323 allow calls without gatekeeper max tries voice network h.323 endpoint registration type . . . . . . . . . . voice network h.323 gatekeeper associate . . . . . . . . . . . . . . voice network h.323 display name . . . . . . . . . . . . . . . . . . . . voice network h.323 rtp port mode . . . . . . . . . . . . . . . . . . . voice network h.323 rtp port base . . . . . . . . . . . . . . . . . . . . Network Dialing Scheme . . . . . . . . . . . . . . . . . . voice destination h.323 endpoint . . . . . . . . . voice destination local channel . . . . . . . . . . voice no destination . . . . . . . . . . . . . . . . . . view voice destination . . . . . . . . . . . . . . . . . voice phone group . . . . . . . . . . . . . . . . . . . voice no phone group . . . . . . . . . . . . . . . . . view voice phone group . . . . . . . . . . . . . . . voice phone group site prefix . . . . . . . . . . . voice phone group site prefix digits . . . . . . . voice phone group type . . . . . . . . . . . . . . . voice phone group format . . . . . . . . . . . . . . voice phone group strip digit length . . . . . . voice phone group forwarding prefix . . . . . . voice phone group forwarding prefix digits . voice phone group add numbers . . . . . . . . . voice phone group delete numbers . . . . . . . voice numbering plan . . . . . . . . . . . . . . . . . voice no numbering plan . . . . . . . . . . . . . . . view voice numbering plan . . . . . . . . . . . . . voice numbering plan activate . . . . . . . . . . . voice numbering plan hunt method . . . . . . . voice numbering plan description . . . . . . . . voice numbering plan destination member . . voice numbering plan phone group member . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . page xii Table of Contents System-Wide VoIP Commands . . . . . . . . . . . . . view voice daughter card . . . . . . . . . . . . . . view voice port . . . . . . . . . . . . . . . . . . . . . view voice channel . . . . . . . . . . . . . . . . . . view voice network card . . . . . . . . . . . . . . voice daughter card statistics collection . . . voice channel reset all statistics . . . . . . . . . view voice channel telephony level stats . . view voice channel telephony channel stats view voice channel voice playout stats . . . . view voice channel dsp stats . . . . . . . . . . . view voice channel error stats . . . . . . . . . . view voice channel modem stats . . . . . . . . view voice channel fax stats . . . . . . . . . . . view voice channel isdn level 2 stats . . . . . voice channel reset telephony level stats . . voice channel reset telephony channel stats voice channel reset voice playout stats . . . . voice channel reset dsp stats . . . . . . . . . . . voice channel reset error stats . . . . . . . . . . voice channel reset modem stats . . . . . . . . voice channel reset fax stats . . . . . . . . . . . . reset voice channel isdn level 2 stats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-259 .5-260 .5-261 .5-262 .5-263 .5-264 .5-265 .5-266 .5-267 .5-268 .5-269 .5-270 .5-271 .5-272 .5-273 .5-274 .5-275 .5-276 .5-277 .5-278 .5-279 .5-280 .5-281 Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .I-1 page xiii Table of Contents page xiv 1 VoIP Overview Introduction This chapter describes Alcatel's H.323 Voice over IP (VoIP) gateway and how telephone or fax calls can be programmed to automatically go through either an enterprise's Virtual Private Network (VPN) via the gateway, and/or the Public Switched Telephone Network (PSTN), a circuit-switched public telephone network that consists of all the interconnected calling networks in the world. Alcatel's H.323 VoIP gateway provides transparent, seamless delivery and connection of local and long distance, inbound and outbound telephone calls initiated through standard PSTN North American (T1), European (E1), and Euro ISDN (Integrated Services Digital Network) digital telephone transmission lines. For specific details on the precise types of calls handled, see Chapter 2, "VoIP Daughtercards" and Chapter 5, "VoIP Commands." As with standard T1, E1, ISDN (Euro) telephone service, VoIP calls can be transmitted fullduplex (simultaneously in both directions). Likewise, Alcatel's H.323 VoIP gateway digitizes phone or fax call signals and uses one of these call transmission services, depending on the type of call, to channel the calls, i.e., carry information to a destination point in the VoIP network. Depending on the configuration of the VoIP network, the calls may also go through the PSTN. For more details on the digitizing process, see Payload Packetization and Digital Signal Processing on page 1-12 for more details. This chapter provides general background information on VoIP networks, clients, gateways and gatekeepers, and includes a list of key features provided specifically by Alcatel's H.323 VoIP network. A VoIP call scenario is illustrated and described along with details on the technologies used in VoIP to explain how calls can be placed over IP. Elements of converged VoIP network are also shown and described, and significant telephone and data communications technologies are explained in relation to the VoIP gateway. Technical standards for the most prominent technologies used in Voice over IP are briefly summarized at the end of this chapter, since as a whole, H.323 ITU standards define the major components using VoIP technologies in network-based VoIP communications systems. VoIP Networks Alcatel's H.323 VoIP gateway for packet-switched IP networks combines the speed, versatility and low cost of IP telephony with standard telephone features for enterprises in North American and Europe (this necessarily entails other continents or countries, such as Mexico, that may have the same requirements). These networks are referred to as VoIP networks. Because data networks as such usually operate with extra carrying capacity (bandwidth), most IP networks are easily able to accommodate voice/fax traffic once the VoIP network is set up. The Internet Protocol (IP) is used mainly because it is supported over many layer 1 and 2 network technologies including Ethernet (10, 100, 1000 Mbps), Token Ring, FDDI and Frame Relay to name a few, including leased lines and satellites. Nearly every router, frame relay device, and network switch used today supports the Internet Protocol. IP delivers any transport media used between local and wide area networks. Enterprise IP networks consist of local area networks (LANs) installed at corporate offices often joined together by corporate wide area networks (WANs). Usually the local area networks support IP on various types of standard data communications technologies such as Ethernet, Token Ring, ATM (Asynchronous Transfer Mode) and FDDI (Fiber Distributed Data Interface). Page 1-1 Introduction Wide area networks are used to support IP connections over leased lines, public frame relay, ATM, satellite and ISDN. At each branch office location, enterprises use routers to connect the remote LANs to the IP WAN. When used with Alcatel's H.323 VoIP gateway, these Virtual Private Networks, or VPNs, allow a portion of the Public Switched Telephone Network to be managed and used by the enterprise. Alcatel's H.323 VoIP gateway provides the Voice over IP network capabilities by means of either digital or analog VoIP daughtercards installed in the switch. The VoIP daughtercards specifically enable enterprises to control the call routing capabilities of their own organizations by using a portion of the PSTN. Getting Started with VoIP Once an enterprise network is ready to provide VoIP using Alcatel's H.323 VoIP gateways, Network Administrators can begin setting up VoIP networks by installing and configuring the appropriate voice switching daughtercard(s). As a whole, Alcatel's VoIP H.323 gateways can be scaled from a minimum of two voice channels per switch to a maximum of 120 voice channels per switch. Switches with the greatest VoIP scalability will use voice switching modules (VSXs) in Omni Switch/Routers. See Chapter 2, "VoIP Daughtercards," for further details on supported configurations and scalability. Initially, an Alcatel VoIP network dialing scheme (AVNDS) must also be selected and deployed via a text-based configuration boot file, wherein each daughtercard must be assigned a unique IP address among other unique gateway identifiers. From that point, operational parameters such as channel and port types can be set using the command line interface (CLI) configuration tool. Comparable text-based (ASCII) configuration boot files may also be quickly generated to configure multiple VoIP-enabled switches with similar requirements. Also, stored in the vsmboot.asc files are voice coding parameters which are pre-configured and kept in profiles. Coding Profiles are configured directly to the components, and define which operational VoIP characteristics will be used, and then implemented according to the instructions contained in the profiles. Coding Profiles consist of general caller information, voice and fax transmission, coding/decoding settings. Preferred Coding Profiles can be automatically selected based upon payload requirements. Coding Profiles are configured at the channel level. VoIP configurations for VoIP callers are established by setting up profiles and then assigning the profiles to each individual H.323 VoIP gateway or daughtercard. Profiles can be created, modified, copied and deleted using one of the available configuration tools. It should be known that in most circumstances, the default settings for the Coding Profiles are sufficient. Additional parameters that require configuration include calling Destinations and Network Numbering Schemes, the latter being comprised primarily of Numbering Plans, Phone Groups and Hunt Methods. Altogether, use of these parameters enable VoIP networks to translate IP addresses from telephone numbers, and allow communications between the VoIP branch offices to be configurable. For more details, see Alcatel VoIP Network Dialing Schemes (AVNDS) on page 1-15. See Chapter 5, "VoIP Commands," for details on using these CLI commands once the H.323 VoIP gateway is configured; refer to this chapter as well if using an optional third-party gatekeeper (server) i.e., NT100 RADVision, on a PC for example, as some additional network parameters will need to be set. For details on configuring the AVNDS, see "Chapter 3, "Network Dialing Schemes" and Chapter 5, "VoIP Commands." For details on installing the cards and setting up VoIP H.323 Gateways, see Chapter 4, "Setup and Installation." Page 1-2 Introduction Alcatel's H.323 VoIP Gateway Key Features Alcatel's H.323 VoIP gateways, which connect voice and data networks, minimize call complexity and dependency on leased telephone lines by allowing enterprises more control over their own call processing. Alcatel's H.323 VoIP gateway is used to transport digitized voice conversations over IP local area networks, which are then sent over wide are networks using such protocols as Frame Relay or ATM. All VoIP daughtercards are compatible with the Alcatel OmniAccess 512 and Omni Switch/Router. As shown below, the following features of Alcatel's H.323 VoIP gateway are supported in this release. · Digital T1/E1 voice and fax transport over IP networks · T1 and E1 telephony interface links to digital Private Branch Exchanges (PBXs) via digital or analog VoIP daughtercards. · T1 robbed-bit Channel Associated Signaling (CAS) · E1 Primary Rate Interface (Euro PRI) and E1 QSIG ISDN Common Channel Signaling (CCS) · E1 Basic Rate Interface (Euro BRI) ISDN; (E1 ETSI) · Foreign Exchange Station (FXS) telephony Loop interface via analog VoIP daughtercard with (FXS) grand-daughtercard (variations includes FX Office-- FXO). · H.323 Network Call Control Gateway (establishes Local Area Network (LAN) terminal links; performs call setup and voice translation functions; provides communications procedures between LANs) · Voice Codecs: Pulse Code Modulation (G.711), Internet Speech (G.723.1), Standard Telephone Quality (G.729A), Realtime Fax over IP (Fax T.38). · Non-Voice Signal Monitoring, Detection and Transmission Protocols: · · · Dual Tone Multi Frequency/Modem Fax Relay Fax Transparency and Fax Relay Modem Transparency and Modem Relay · PSTN Fallback via Deadman Relay Switch · The H.323 VoIP gateway is capable of providing PSTN fallback for VoIP calls in the event of a power failure in the VoIP network by means of a Deadman relay switch on the digital VoIP daughtercards. For more information on the Deadman switch, see Chapter 2, VoIP Daughtercards." · Echo and jitter controls on digital VoIP daughtercards. · Pre-configured, modifiable AVNDS (Alcatel VoIP Network Dialing Schemes) with corresponding text-based (ASCII) configuration boot files (vsmboot.asc files). · VoIP Text-based Command Line Interface (CLI) configuration tool. o Note o When used separately, the terms E1 and ETSI both entail European PRI and BRI interfaces. E1 ETSI used together as one term refers specifically to Euro PRI. Page 1-3 VoIP Telephone Calls VoIP Telephone Calls H.323 VoIP telephone calls, which can carry either voice, facsimile, or modem transmissions over IP networks, are switched to the packet-based network and connected to the calling destination (an IP device) via a unique IP address and local/remote dialing plan (actually two Alcatel VoIP Network Dialing Schemes rolled into one). The numerical IP address, also serving to identify calls intended for VoIP networks, is determined and translated from a destination telephone number in a phone directory database while it is being entered, and the call is in progress. (It should be noted here that callers do not need to remember the IP addresses, only the called party or destination phone number). See Chapter 3, "Network Dialing Schemes," for more information on the AVNDS. H.323 VoIP telephone calls are transparent so callers don't have to worry about any special procedures, except being aware of a dialing plan that may require them to dial a prefix, such as 7, before a call can be placed across the VoIP network. This would be similar to current dialing plans requiring callers to dial 9 before an office call can be placed (9 is the prefix most often used by PBXs to access the PSTN). VoIP calls initiated from standard telephone handsets after a preset number of digits are dialed, for example, can be immediately transmitted using IP data networks whereby digital or analog signals, meant to set up connections for carrying information, are intercepted by Alcatel's H.323 VoIP gateways in the network. These gateways translate the phone numbers into IP addresses, convert the information to digital packet form, and then deliver the calls over the network and the PSTN as shown below. A VoIP Call Scenario Once a VoIP network is set up a typical VoIP call scenario might go something like this. Local Telephone Number Remote Telephone Number VPN Step Call setup begins 1 Source IP Address VoIP H.323 Gateway VoIP H.323 Gateway Destination IP Address PBX #1 PBX #2 Dial Tone Dial Destination Number VoIP Call Scenario -- Step 1: Call Setup Page 1-4 VoIP Telephone Calls Local Telephone Number Remote Telephone Number VPN Step Session setup with remote gateway Dialed digits translated to IP address Source IP Address VoIP H.323 Gateway VoIP H.323 Gateway 2 Destination IP Address PBX #1 PBX #2 VoIP Call Scenario -- Step 2: Call Progress As the caller dials, the H.323 VoIP gateway collects the dialed digits and then ultimately translates the digits using a pre-configured Numbering Plan and Phone Group into the IP address. A VoIP session is then initiated with the remote gateway (when gatekeeper not used). After the gateways determine that the VoIP call can be placed across the IP network, the gateways negotiate call capabilities using preconfigured coding profiles, and then optionally strip before sending the extension digits from the local to the remote gateway where they are delivered either to the phone, PBX, or keyset. The call can be processed as either a local or long distance call depending on how the remote gateway is configured. A ringing or busy signal is transmitted to the caller once the call is connected. If the call is answered, the gateway sends the voice or fax transmissions. If the wide area network is unavailable, calls may not go through, in which case callers receive a busy signal. When a caller hangs up the receiver, the VoIP call session is terminated. Multiple gateway trunks may be used for all calls except those initiated from keysets which must go directly to the gateway. Page 1-5 VoIP Telephone Calls Local Telephone Number Remote Telephone Number VPN Step Call setup completed (message returned) 3 Source IP Address VoIP H.323 Gateway VoIP H.323 Gateway Destination IP Address PBX #1 PBX #2 Phone Rings IP call VoIP Call Scenario -- Step 3: Call Setup Completed Local Telephone Number Remote Telephone Number VPN Step Call answered (message returned) 4 Source IP Address VoIP H.323 Gateway VoIP H.323 Gateway Destination IP Address PBX #1 PBX #2 IP call VoIP Call Scenario -- Step 4: Remote Call Answered as VoIP Call Page 1-6 Elements of a Converged Network Elements of a Converged Network Alcatel's H.323 VoIP gateway is based on a complex, dual-technology infrastructure taken from what have been in the past two fairly distinct industries -- namely, Telecommunications (a.k.a. Telephony) and Data Communications. It converges voice and data into enterprise, Internet Service Providers (ISPs), and carrier networks to provide various levels of VoIP services using intelligent switches in order to generate long-term cost reductions for telephone services between sites. The standard or key elements of a converged H.323 VoIP network are described below and shown in the illustration Elements of Converged Voice/Data Packet-Switched VoIP Network and Circuit-Switched PSTN on page 1-9. This illustration is intended to provide a sample, visual representation of all the various devices that may be used in a VoIP network and how they may interconnected. VoIP network interoperability is based on ITU H.323 network call control standards and multiple vocoder support. See also abbreviated International Telecommunication Union (ITU) Standards on page 1-17. By means of either digital or analog VoIP daughtercards installed in Alcatel switches, the basic elements required for providing enterprise H.323 VoIP gateways in packet-switched IP networks are readily accommodated, including the client, the gateway, and the gatekeeper as described. VoIP H.323 Client The Client is the device initiating and/or receiving the call. This can be a standard telephone handset or some other H.323 VoIP-capable device in an IP network. VoIP H.323 Gateway Alcatel's H.323 VoIP Gateway is the device used to make the transition from the packetized voice network to a circuit-switched network, e.g., PSTN, and back. Functionally, the enterprise VoIP gateway is comprised of voice to IP network converter components, .e.g, DSPs, on the voice switching daughtercards. In VoIP, the process for call placement is the same as in a service provider system except that the gateway is accessed from Customer Premise Equipment (CPE) instead of from a local service provider, e.g. CLEC (Certified Local Exchange Carrier). o Note o PBX and Key Systems setup, installation and configuration procedures are beyond the scope of this manual. Gateway devices intercept then direct electric signals between networked devices. With VoIP, gateways translate transmission formats between voice CPE and H.323 IP network call control endpoints and terminals, including communications procedures between gateways. They also translate between codecs, perform call setup/teardown on LANs and on circuit-switched telephone networks. Gateways are entrance and exit points into VoIP networks that without hardwiring perform code and protocol conversions, as well as signal filtering. VoIP gateways contain a user-definable phone directory database of phone number to IP address mappings; this is called an Alcatel VoIP Network Dialing Scheme (AVNDS). See Chapter 3, "Network Dialing Schemes," for details. Modifications to the local phone directory database are downloaded through the IP network to the switch, and may be accessed using the VoIP configuration interface. The phone directory database is built as the VoIP network is configured, and is contained in the VoIP configuration boot file (vsmboot.asc). Numbering plans, phone groups and destinations as part of the AVNDS comprise a portion of the phone directory database used by Alcatel's H.323 VoIP gateway. Page 1-7 Elements of a Converged Network Gateways are considered H.323 terminals or H.323 endpoints in H.323 IP networks. Terminals are also the endpoints where telephone lines connect to network circuits. Terminals provide real time, two-way communications for local area network (LAN) endpoint destinations. All terminals as such must support voice communications and H.245 in-band call controls to use and negotiate channels. See also abbreviated International Telecommunication Union (ITU) Standards on page 1-17. VoIP H.323 Gatekeeper (Optional) The H.323 Gatekeeper (server or workstation) is the device that verifies client VoIP privileges and translates telephone numbers into IP addresses. It should be noted that H.323 gatekeepers are not required to use Alcatel's H.323 VoIP Gateway. In lieu of an H.323 VoIP gatekeeper, Alcatel's H.323 VoIP gateway uses its patent-pending Alcatel VoIP Network Dialing Scheme (AVNDS) to perform IP address translations. o Note o Gatekeeper setup, installation and configuration procedures are beyond the scope of this manual. Alcatel recommends and has tested extensively use of Alcatel's H.323 VoIP gateway with the NT100 RADVision Gatekeeper. Gatekeeper devices identify, track and control traffic flowing through them, and perform other functions such as gateway registration, admission and bandwidth controls. Page 1-8 Elements of a Converged Network VOICE Gatekeeper LAN Digital Clients Microsoft NetMeeting Ethernet Ethernet IP Address 4400 Ethernet Client OmniPCX Packet-Switched VoIP Network Central Site WAN H.245 VPN WAN H.245 T1 Remote Site E1 BR I VoIP H.323 Gateway WAN VoIP H.323 Gateway BRI Telephone PSTN Fallback (Deadman Relay Switch) Euro ISDN PBX #1 Circuit-Switched ISDN PSTN NO. AMER. PSTN Key System POTS/ PSTN Analog FX FX O/ S VoIP H.323 Gateway Elements of Converged Voice/Data Packet-Switched VoIP Network and Circuit-Switched PSTN Page 1-9 H.323 VoIP Gateway Voice and Convergence Features H.323 VoIP Gateway Voice and Convergence Features As shown below, the main functions handled by Alcatel's H.323 VoIP Gateway include the following: · · Telephony Signaling -- used to communicate with the PSTN or Customer Premises Equipment (CPE). Payload Packetization and Digital Signal Processing (via DSP) -- converts PCM voice packets from circuit-switched network to H.323 packets on IP network and the reverse. H.323 Network Call Control -- handles H.245 and H.225 packet processing, e.g., connect, disconnect. Alcatel VoIP Network Dialing Schemes (AVNDS) -- handles conversions between phone numbers and IP address of H.323 devices. Network Switch Backplane Interface -- connects H.323 VoIP gateways to switch and ultimately to IP network. · · · These functions can generally be divided into either voice or convergence features, based on the controls they provide over VoIP in the switch. For the most part, the voice features include separate controls for signaling and for voice interoperability, whereas, the convergence features encompass H.323 call control and voice/data interoperability via the use of AVNDS in IP networks. Alarms VoIP Network Call Control (H.323) Signaling Alarms Telephony Signaling (Digital or Analog) Control Packets PSTN, PCX/PBX, BRI phone DTMF (Digit Collector) Voice Ports Configuration Alcatel Voice Network Dialing Schemes (AVNDS) Network Switch Backplane Interface Switch Bus Daughtercard Activation Alarms Payload Payload Packetization (Voice, Fax, Modem) RTP Payload Packets VoIP Daughtercards and Enterprise VoIP Features Page 1-10 H.323 VoIP Gateway Voice and Convergence Features Signaling Control and Voice Interoperability (Voice Features) The ability to accommodate voice traffic using VoIP switches installed in data networks is achieved by means of signaling controls and voice interoperability features. The VoIP signaling control and voice interoperability functions includes telephony signaling and payload packetization as described. Telephony Signaling Telephony signaling is used for signaling with telephone equipment, e.g., PBX, via the telephony interface, as well as to control the communication signaling between the H.323 VoIP gateway and the Customer Premises Equipment (CPE). It detects the presence of new calls, collects dialed digit information (telephone number in some form or another) entered by the caller to route a call via an AVNDS to its destination point, and is also used to detect the end of calls (off hook). Telephony signaling provides call progress supervision by generating supervisory and call progress tones, as well as DTMF (Dual Tone Multi Frequency) tones for outbound calls. It also provides DSP (Digital Signal Processor) interfacing control, and transfer of PCM-based voice packets to and from the DSP subsystem or DIMM (DSP Interface Management Module). It coordinates with the DSPs to select voice coders (codecs) at startup when a particular vocoder is needed. When a call is received, telephony signaling is responsible for opening channels and PCM data streams to the DSPs to process the voice data. The signaling controls provided by the Telephony Signaling functions includes the following: · · · · · · Call Progress Tone and Tone Detection -- detects individual in-band frequencies and converts them into tones or other signaling events, e.g., answer or busy signals. Dialing Timers -- used to time incoming signaling events, e.g., how long to wait for a wink start, or how long to wait for another digit. E&M Signaling (Common, Wink Start, Immediate Start and Delay Start) -- customizes attribute settings or parameters to match CPE. (Available only on VSD-T1.) Foreign Exchange Station (FXS) Loop Start -- customizes attribute settings or parameters to match CPE. Foreign Exchange Office (FXO) Loop Start -- customizes attribute settings or parameters to match CPE. Caller ID -- looks for Caller ID information, e.g., calling party telephone number between first and second ring. The telephony signaling configuration options for telephone signaling interfaces, e.g., ring delay and cadence (ringing rhythm), are assigned to the physical ports on the daughtercard, including the T1 and E1 line specifications. All options are defined at the channel level. Parameters for telephony signaling and VoIP network preferences are pre-configured in textbased configuration files referred to as vsmboot.asc files. The parameters are stored in the boot files and subsequently assigned to a daughtercard and/or its components. For more details on setting these parameters, see also Chapter 5, "VoIP Commands." Page 1-11 H.323 VoIP Gateway Voice and Convergence Features Payload Packetization and Digital Signal Processing Payload packetization is responsible for conversion between time-continuous telephony (analog or digital payload) at the telephony interface and Real Time Protocol (RTP) packets on the data network interface. It supports voice compression, echo cancellation, Fax and DTMF Relay (demodulation/modulation), modem data transport (up to 14400 baud), voice activity detection and comfort noise generation, as well as packet arrival de-jittering. Physically, the payload packetization function is implemented on the DSPs (DIMMs), with control and configuration on the Motorola MPC860 processor. Configuration is performed through the vsmboot.asc file on the switch. Upon VoIP daughtercard activation, the configuration is transferred from the switch to the daughtercard. See Chapter 4, "Setup and Installation," for more information. The controls for voice interoperability provided by the payload packetization functions include the following: · · · · · Codecs (see also Coding Profiles -- H.323 Call Capabilities) -- provides encoding/ decoding of H.323 packets. Voice Echo Cancellers -- reduces echo on voice conversations. Fax or Modem over IP -- allows fax/modem calls to be transmitted via H.323. Voice Activity and Silence Detection -- detects voice conversation (or lack thereof) to reduce H.323 bandwidth requirements. Comfort Noise and Jitter Buffer -- generates slight background noise (white noise) on the voice conversation, so callers do not think the connection has failed. Digital Signal Processors, or DSPs as they are more commonly known, are math-intensive coprocessors used to convert and manipulate information, especially in telecommunications systems (systems that transmit all types of data including voice and video). They are also programmable chips, well-suited for VoIP as DSPs have the ability not only to convert but to compress analog signals into various digital formats, i.e., perform digital signal processing. Although DSPs do not have any direct analog input/output since they are actually digital devices, they can accept digitized analog data rather than raw analog signals. As a result, DSPs are used in the digital and analog VoIP daughtercards developed by Alcatel to bring switch-enabled VoIP to enterprises; however, before the digitized and compressed voice signals can be delivered as "voice data" in a VoIP network, they must be packetized into H.323 packets. Packetized voice is digitized voice compressed into finite bit stream of IP packets, that carry the "voice payload" between remote and distant locations, across the IP network and make processing VoIP calls in IP networks possible. Once compressed and packetized, periodic delays (jitter) to make the call sound smoother must be imposed on the transmission of these packetized "voice" conversations to mimic "real time voice" (resonating by nature in continuous "analog" waveform). DSPs are used further to reduce the delays from conversion and compression to ensure quality voice communications without affecting the real time voice processing and compression that occurs simultaneously. To transmit the compressed data (digitized voice) across the IP network, the Real Time Protocol (RTP) is used. RTP streamlines and then transports voice packets, including interactive multimedia packets over IP, although it does so without any guarantees or quality of service provisioning. o Note o H.323 VoIP telephone calls automatically receive the highest priority in the VoIP network via the Quality of Service ToS bit. For more information, see the switch manual. Page 1-12 H.323 VoIP Gateway Voice and Convergence Features Voice packet transmissions, or the "payload," are expedited by engaging the User Datagram Protocol (UDP) for faster delivery, packets which by necessity include the IP network call transport header information. Resultant jitter caused by delays imposed on the payload packets upon arrival to their destinations is also handled by the DSPs. Layer 2 Header IP Header UDP Header RTP Header Voice/Fax Data Payload Voice Packet Transmission UDP is needed by RTP to keep pace with "Real Time Voice" but lacks controls and error checking capability. DSPs can monitor calls in progress, detect voice activity and handle echo cancellation (the filtering of unwanted transmission signals as specified in ITU algorithm standards G.160 and G.126); comfort level background (white) noise can also be generated on either the transmitting or receiving end. Since digital signal processing affects nearly every operation in VoIP, numerous DSPs are incorporated adjacent to the supporting MPC860 CPU signaling controller in the voice switching daughtercards (normally used with voice switching modules), comprising the core of Alcatel's enterprise VoIP on the call processing end. The DSPs and the Motorola MPC860 controlling processor work in unison to support the various protocols and interfaces that implement the enterprise VoIP telephony functions contained in software on the voice switching daughtercards. In a nutshell, the DSPs are the voice processors, and the MPC860 controller is the data communications processor on the daughtercards. Altogether, the above components provide T1, E1 and ISDN voice and data synthesis processing, with scalable versions of each bringing enterprises any-to-any switching functionality that now, with enterprise VoIP, includes least-cost call routing for VoIP Virtual Private Networks (VPNs). Signal Recognition Initially, digital signal processing involves DSP detection of an array of voice signaling types using Channel Associated Signaling (CAS) repetitive circuit-state signaling protocols (for T1 and E1 lines). Many forms of call signaling exist to set up and end calls, most of which result in the ringing of a phone or connection of a fax machine. These forms entail newer line signaling methods that use digital pulses (PCM, or Pulse Code Modulation), analog touchtones such as DTMF (Dual Tone Multiple Frequency), and other much older analog signals in all their assortments, including but not limited to: Ear & Mouth (E&M), Loop Start, Ground Start, Foreign Exchange Subscriber (FXS) and Wink Start. Each signaling method was developed through the years by the telephone industry to provide Plain Old Telephone Service (POTS). E&M signaling, of which there are five interface types, is the most widely used method for connecting calls to PBXs, telephone switching systems which use channelized T1 or E1 lines to transmit signals and multiplex digitized voice. T1 robbed bit signaling is an example of narrow or in-band signaling -- where signaling tones are passed along the same circuit as someone's voice. ISDN (Integrated Services Digital Network), on the other hand, is another type of signaling wherein voice transmissions are digitized then placed on separate broad or out-of-band channels (so signaling tones are not passed along the same circuit as someone's voice). This prevents signaling or other intrusions into the calls, and usually provides faster transmission. ISDN is a common protocol in the Common Channel Signaling (CCS) network architecture used for exchanging information between out-of-band signaling networks and telecommuniPage 1-13 H.323 VoIP Gateway Voice and Convergence Features cations nodes in the network. ISDN does not use T1 (or DS-1) robbed bit signaling, where bits are taken from voice data to carry signaling. Alcatel H.323 gateways support in-band and out-of-band signaling. Encoding Once signaling types are determined they are analyzed and converted by the appropriate DSP voice coder (vocoder) into digital signals, which are ultimately converted and expanded back (re-modulated) into real voice. More specifically, after signal recognition and analysis, DSPs convert (encode) the amplitude of incoming analog signals into digital form using codecs, or CODer/DECoders. The basic encoding schemes, or companding methods in use today, are for PCM which "encodes" analog signals into digital signals. Although the PCM companding methods used for T1, which follows Mu-Law, and E1 which follows A-Law, differ mainly in their algorithms, their purpose is much the same. (Companding is a contraction for compression and expansion.) However, A-Law and Mu-Law are incompatible. They use different methods, for example, to sample analog signals. Next, the digitally encoded signals are compressed using industry standard vocoders. These are devices that use speech compression/decompression algorithms to analyze and convert analog waveforms into digital signals and reduce related bandwidth requirements. DSP Digitizer Compressed Vocoder Real Time Protocol Encapsulation G.711 G.723.1 G.729 G.729a Analog Digital Voice/Data Encoding and Call Compression Compression The appropriate vocoder used for VoIP calls is then negotiated by the H.323 VoIP gateway prior to call placement. As an added bonus, but with some variations in protocols, the same DSP technology that is used for voice compression also works with fax modems. (For that reason, it can be assumed that references to voice signal packets inherently include "fax" packets.) The codecs and vocoders used in enterprise VoIP adhere to the ITU recommendations that fall under the H.323 IP network call control umbrella of interoperability standards for multimedia communications over packet-switched local area networks (part of the Series H Recommendations for Audiovisual and Multimedia Systems). The ITU's H.323 suite of specifications includes the H.245 in-band call control specifications. For the signaling vocoders (G.711, PCM; G.723.1, Internet Speech; G.729/G.729a; Standard Telephone Speech), the algorithms in the Series G Recommendations for Transmission Systems and Media, Digital Systems and Networks are used. Page 1-14 H.323 VoIP Gateway Voice and Convergence Features H.323 Call Control and Network Interoperability (Convergence Features) The ability to accommodate voice (also fax and modem) traffic compressed into data form via payload packetization for transport across data networks is achieved through the use of H.323 call controls and Alcatel VoIP Network Dialing Schemes (AVNDS) as described. H.323 Network Call Control H.323 network call controls are responsible for the procedures and protocols necessary to establish/tear down VoIP calls across the IP network. The VoIP gateway implements the H.323 network call control standards, which include the following: · · The H.225/Q.931 protocol that performs call establishment and tear down by establishing a reliable call signaling channel. The H.245 protocol that establishes a reliable H.245 in-band channel for communications between all endpoints or terminals, i.e., gateways, for capability exchange and other messages. The registration, admission and status (RAS) protocol that creates a RAS channel to carry RAS messages between an endpoint and a gatekeeper. The H.323 IP network call control standards that support multimedia communications over local area networks. · · The H.323 network call controls and capability provided by the H.323 call control functions includes the following: · · · H.323 gateway (VoIP Switch) H.323 gatekeeper (e.g., RADVision Server) (Discovery, Configuration and Operation) H.323 call capabilities (Coding Profiles or Codecs, Voice Network Delay Buffers) The voice network configuration options include general network information, H.323, H.225, and H.245 configuration settings, gateway, gatekeeper and registration parameters, and Real Time Transport Protocol (RTP) session parameters that must be specified for IP network communications. Voice network call control parameters are configured at the VoIP daughtercard level. Alcatel VoIP Network Dialing Schemes (AVNDS) The AVNDS are responsible for the operations and configuration of the VoIP daughtercard and/or voice switching module, e.g, VSX. AVNDS are implemented on the Motorola MPC860 processor, and the switch. AVNDS are responsible for providing the interface to configure and maintain all VoIP daughtercards (H.323 gateways) on the entire VoIP network. Additionally, both standard packet Management Information Bases (MIBs) and proprietary voice packet MIBs are supported. The AVNDS are used to store information contained in VSM (Voice Switching Module) configuration boot files (vsmboot.asc) concerning the configuration of the VoIP network, particularly the following: · · · Destinations (H.323 endpoints, H.323 local channel destinations) Phone Groups (e.g., strip digits and extensions) Numbering Plans (hunt methods and hunt groups) Page 1-15 H.323 VoIP Gateway Voice and Convergence Features The AVNDS handle inbound/outbound calls routing to/from the VoIP network and local ports. AVNDS are also used to set up calls and translate IP addresses to telephone numbers, and can be used with, or in lieu of, H.323 VoIP gatekeepers. VoIP configuration boot files and profiles simplify VoIP configuration of Alcatel's H.323 VoIP gateways (VoIP daughtercards) by using sets of pre-configured parameters that can be assigned to the various manageable components. Various configuration elements, e.g., profiles, have a user-defined name associated with it. VoIP daughtercard configurations are stored in the switch. Destinations which consist of remote network and local calling gateways, including H.323 gatekeepers, allow Network Administrators to configure a destination IP address and its specific protocol. Local channel destinations are considered subdestinations. Destinations, which are appended to hunt methods, are configured at the daughtercard level. Phone groups are used to indicate what telephone numbers are available. They also define digits to be stripped and forwarded. Phone groups are configured at the daughtercard level. Voice numbering plans use hunt methods to arrange telephone lines so that when calls come into the network they will ring in a certain order. For example, to use PSTN fallback, all phone groups must be set up with the last group element indicating the local destination or gateway to fall back on when a call cannot be placed over the VoIP network. Hunt methods in voice numbering plans are configured at the daughtercard level. Hunt methods dictate what to do if the first line tried is busy, i.e., hunt methods are used to track down lines in a certain order until an available line is located. Phone line destinations can be grouped as desired in user-defined groups, such as by divisions or departments, location, or some other meaningful grouping. For information on setting up and using the AVNDS (Alcatel VoIP Network Dialing Schemes), see Chapter 3, "Network Dialing Schemes," Chapter 4, "Setup and Installation," and Chapter 5, "VoIP Commands." Switch Backplane Interface The switch backplane interface is responsible for the payload packet transport, and VoIP daughtercard management message transport between the VoIP daughtercard and the host switch. Physically, the interface consists of a 100-pin connector between the VoIP daughtercard and the motherboard. All functions of the H.323 VoIP gateway are implemented on the MPC860 controllers on the daughtercards, the VSX (OSR configurations only) and the switch. Page 1-16 VoIP Standards for Development VoIP Standards for Development Alcatel's H.323 VoIP gateway is designed to function in accordance with the following IP Telephony and Internetworking standards currently available as briefly summarized below. International Telecommunication Union (ITU) Standards ITU-T technical, operational, and tariff recommendations are used for standardizing telecommunications on a worldwide basis. ITU H.323 IP network call control standards apply to VoIP. These standards define the major components, namely, Telephone Terminal Equipment, Gateways, Gatekeepers and Multipoint Control Units (MCUs) for H.323­based communications systems: Series H.323 -- Series H: Audiovisual and Multimedia Systems (Infrastructure of audiovisual services -- systems and terminal equipment for audiovisual services). H.323: this standard is specifically concerned with recommendations for real time audio, video and/or data and facsimile transmissions over H.323, Packet­based Multimedia Communications Systems. In reference to Alcatel's enterprise VoIP, the series as a whole relates to gateway devices used in VoIP to handle audio, video, data and facsimile transmission over IP or packet networks. The standard specifically includes the newer ITU recommendations for the Internet facsimile protocol (T.38) that is used to 1) exchange messages and data between facsimile gateways connected via an IP network, and 2) message transport (depending upon bandwidth availability) using either TCP/IP or UDP/IP network protocols. (T.38 is incorporated into the first release of Alcatel's enterprise VoIP). Series H.225 -- Series H: Transmission of Non­Telephone Signals (Infrastructure of audiovisual services -- Transmission multiplexing and synchronization. H.225: this standard is specifically concerned with recommendations for narrowband visual telephone services defined in H.200/AV.120-Series transmission paths for local area networks (LANs) providing non-guaranteed quality of service (QoS) which is less than that of ISDN PRI protection and recovery mechanisms. This recommendation describes how non-guaranteed QoS LANs provide conversational services for audio, video, data and control information in H.323 equipment. The series relates to gatekeeper devices used in VoIP to provide services across LANs. Gatekeepers are centralized network devices performing IP address translations and bandwidth management. Alcatel's H.323 VoIP gateway uses AVNDS to work with, and in lieu of, gatekeepers. (Note: Standard includes codec support for call synchronization.) Series H.245 -- Series H: Audiovisual and Multimedia Systems (Infrastructure of audiovisual services -- Communications procedures), Control protocol from multimedia communication. H.245: this standard specifies syntax and semantics of terminal information messages, particularly receiving and transmitting capabilities, mode preferences from the receiver, including logical channel signaling, Control and Indication messages. Signaling acknowledgements are specified to ensure reliable audiovisual and data communications. The series relates to Multipoint Control Units used in VoIP to provide signaling and coding for call synchronization. Codec Support (G.711, G.723.1, G.729a) G.711 (PCM Encoding) This is the ITU recommendation for an algorithm designed to transmit and receive A-Law and Mu-Law PCM voice at digital bit rates of 48, 56, and 64 Kbps. It applies to digital telephone sets on digital PBS (cellular) and ISDN channels. Support for this algorithm is required for ITU-T compliant videoconferencing (the H.320/H.323 standard). Page 1-17 VoIP Standards for Development A-Law and Mu-Law are processes needed to compand digital signals. A-Law is used in most countries except for the U.S., Canada and Japan where Mu-Law is more common. Companding is the process of compressing the amplitude range of a single signal, and then expanding them at the receiving end back to their original form. Although it is impossible to exactly reproduce an analog signal digitally, companding greatly improves the accuracy of this process. PCM uses two different companding processes. For this reason, PCM A-Law is used for international networks. A-Law: The PCM coding and companding standard used in Europe and in areas outside of North America. A-Law encoding samples audio waveforms used in the 2.048 Mbps, 30-channel PCM system (E-carrier). Mu-Law (E-Law): The PCM voice coding and companding standard used in Japan and North America. A PCM encoding algorithm where analog voice signals are sampled 8,000 times per second with each sample represented by an eight-bit value, and a raw 64 Kbps transmission rate. All sample bits are inverted before transmission. o Note o A-Law and Mu-Law are incompatible. For example, a signal sent with A-law cannot be received by a system using Mu-Law. G.723.1 This is the ITU-T algorithm recommendation used for compressed digital audio over Plain Old Telephone Service (POTS) lines. It is the voice part of H.324 (POTS video conferencing). This algorithm runs at 6.3 or 5.3 kbps (20 bytes per 30ms interval) and uses linear predictive coding and dictionaries, which help provide smoothing. The smoothing process is CPU-intensive during real time based activities. G.729a This is the ITU's standard voice algorithm ­ CS-ACELP (Conjugate Structure Algebraic Code Excited Linear Predictive for the encoding/decoding of speech at 8 Kbps using conjugatestructure, algebraic-code excited linear predictive method. G.729 is supported by inter alia (among other things), American Telephone and Telegraph, France Telecom and Japan's Nippon Telephone and Telegraph. VON (Voice on the Net) Developments The VON (Voice on the Net) Coalition is concerned with developments in Internet Telephony and IP Telephony around the world. It is an incorporated, non-profit U.S. organization working with the government, business and other groups and individuals on regulations that affect this technology and its use. Alcatel's H.323 VoIP gateway was designed with the considerations of the VON Coalition in mind. Compression techniques and DSPs improve the quality of VON transmissions and minimize problems associated with IP packet delays. Page 1-18 VoIP and VLANs VoIP and VLANs Alcatel VoIP (VSD, VSB, VSA) modules cannot not be in a Virtual LAN (VLAN) with non-voice ports (i.e. data ports), IP phone ports, etc. All voice traffic must route in and out of the VoIP VLAN. Page 1-19 VoIP and VLANs Page 1-20 2 VoIP Daughtercards Introduction This chapter describes the voice switching daughtercards that can be installed in Alcatel switches to provide H.323 VoIP gateways in VoIP networks. Using ITU H.323 IP telephony standards, the H.323 VoIP gateway converts telephone or fax calls between the circuit switched Public Switched Telephone Network (PSTN) and packet-switched VoIP networks. Alcatel's H.323 VoIP gateways are typically used to handle VoIP calls as such placed across local and wide area networks between branch offices in remote enterprises, although the gateways are suitable for use in carrier applications, too. See Chapter 1, "VoIP Overview" for a more in-depth description of Alcatel's VoIP H.323 gateway operations. Different VoIP daughtercards, as described below, are required depending on the telephony interface required to transmit and receive calls in the VoIP network. Furthermore, to digitize the VoIP calls, the daughtercards utilize digital signal processors (DSPs) containing a specified number of channels, which in turn determine the maximum number of calls that can be placed at one time on the card. The VoIP daughtercards are referred to as voice switching daughtercards and, when installed in the switch, they are sometimes referred to as Voice Switching Modules (VSMs). Currently, the VoIP daughtercards can be installed in either the OmniAccess 512 or Omni Switch/Router. A blade installed in an OSR containing either one or two VoIP daughtercards of the same type is referred to specifically as a VSX or VSX switching module (for details see VSX Switching Module on page 2-25). For information on configuring either of these switches, refer to the appropriate switch user manual. This chapter also depicts the port pinouts and jumper settings for all of the VoIP daughtercards where necessary, as well as the Deadman switch, and Cross-Over toggle switches available on certain voice daughtercards. The front panels for the VoIP daughtercards, including the front and bottom views of the cards, are shown to illustrate certain components relative to important operations of the H.323 VoIP gateway in the switch. All VoIP daughtercards can be field-installed. For details on installing the cards, see also Chapter 4, "Setup and Installation." For details on configuring the switch to run VoIP, see Chapter 3, "Network Dialing Schemes" and Chapter 5, "VoIP Commands." Page 2-1 Introduction VoIP Daughtercard Types There are two types of VoIP daughtercards: digital and analog. The digital voice switching daughtercards includes the VSD daughtercard used for digital calls placed in either North America and/or Europe, and the VSB daughtercard used specifically for digital calls placed in Europe. The analog voice switching daughtercard (VSA) is used only in North America for placing analog POTS (Plain Old Telephone Service) calls, e.g., to the PSTN. The basic VoIP daughtercards, which allow the switch to make these various types of phone connections, are listed and described below. · · · VSDs (T1, or E1 QSIG and E1 ISDN PRI Digital) (North America and Europe) VSBs (Euro BRI ISDN Digital) (Europe) VSAs (Analog) (North America and Europe) VSD -- The digital voice switching daughtercards (VSDs) have two physical port connections which can be either T1 or E1 (called Dual T1 or Dual E1). Associated with each of the digital physical ports there can be either 48 channels for T1 connections, or 60 channels for E1 connections. The VSD card supports the following protocols or voice port interface connections for VoIP networks in North America: T1. The VSD card supports the following protocols or voice port interface connections for VoIP networks in Europe: E1 (QSIG) or E1 ISDN PRI (QSIG is another name for ITU Q.931). The VSD card does not support the following protocols: T1 ISDN PRI, T1 QSIG or Euro BRI ISDN (E1 ETSI). The VSD-60CH T1/E1 card is considered a high-end VoIP daughtercard as it provides the most channels. Reliable non-digitized voice processing is available only between two ports of the same interface type on a single daughtercard, and not between daughtercards. See Voice Switching Daughtercard -- Digital on page 2-6 for more details. For more information on the voice port interface types for the digital VoIP daughtercards see also the digital port configuration commands in Chapter 5, "VoIP Commands." VSB -- The other type of digital voice switching daughtercard is normally referred to as a VSB since it provides Euro BRI ISDN (E1 ETSI) protocol or interface port connections for VoIP networks in Europe. It differs mainly in that it has four ports, each with two (B) bearer channels and one (D) data channel. B-channels carry voice and data content, whereas D-channels are dedicated to carry control signals or call processing data for the B-channels. Each B-channel contains one DS0 voice channel. The VSB card does not support the following protocols: T1 ISDN PRI or T1 QSIG; T1 or E1 QSIG; E1 PRI or T1 BRI. See Voice Switching Daughtercard -- Euro BRI ISDN on page 2-13 for more details. VSA -- The analog voice switching daughtercard (VSA) can contain an even number of analog ports from two to 16 depending on whether the card provides Foreign Exchange Station (FXS), e.g., telephone set (TelSet), or Foreign Exchange Office (FXO), e.g., Central Office (CO) port interface connections in an OmniAccess 512 or Omni Switch/Router. Each analog FXS port allows the connection of one off-the-shelf TelSet, or some other voice device, e.g., analog fax machine, analog phone answering machine, whereas each analog FXO port allows the connection of an FXO cable to a wall outlet or CO). SeeVoice Switching Daughtercard -- Analog on page 2-19 for details. o Note o When used separately, the terms E1 and ETSI both entail European PRI and BRI interfaces. E1 ETSI used together as one term refers specifically to Euro PRI. Page 2-2 Introduction The table below shows the basic versions of the VoIP daughtercards, all of which were designed with various configurations in mind to fully support the wide range of features used in Voice over IP. See also Chapter 4, "Network Dialing Schemes," for more configuration details. Also, not all configurations shown below may be currently available for purchase. Voice Daughtercard VSD VSD-12CH VSD-24CH VSD-36CH VSD-48CH VSD-60CH VSB VSB Description Two (2) digital T1/E1 RJ-45 voice ports, 12 compressed voice channels Two (2) digital T1/E1 RJ-45 voice ports, 24 compressed voice channels Two (2) digital T1/E1 RJ-45 voice ports, 36 compressed voice channels Two (2) digital T1/E1 RJ-45 voice ports, 48 compressed voice channels Two (2) digital T1/E1 RJ-45 voice ports, 60 compressed voice channels Four (4) digital E1 (Euro BRI) RJ-45 voice ports, TE (Terminal Equipment), NT (Network Terminator), Point to Point or Point to Multipoint NT, 8 compressed voice channels Four (4) analog RJ-11 voice ports Eight (8) analog RJ-11 voice ports Two (2) analog RJ-11 voice ports Four (4) analog RJ-11 voice ports Four (4) analog FXS and (2) analog FXO RJ-11 voice ports VSA-FXS VSA-4FXS VSA-8FXS VSA-FXO VSA-2FXO VSA-4FXO VSA-MIX VSA-4FXS2FXO Page 2-3 Introduction Digital Signal Processors (DSPs), DIMMs and Available Channels All digital VoIP daughtercards have four vocoder channels available per DSP chip. DSPs are scalable in increments of three (in DSP DIMM modules) to better accommodate the needs of a VoIP network, and to reduce costs since the number of DSPs required is based on the number of simultaneous vocoder channels needed. DIMM stands for DSP Interface Management Module (not Dual Inline Memory Module). Voice switching daughtercards, such as the VSD T1/E1 card, can contain up to 15 digital signal processors running at 100 MIPS (millions of instructions per second), providing voice processing functions for up to 60 DS-0s, or 60 Digital Service 0 channels (24 DS-0s are equal to one DS-1, or T1, channel). The number of populated DIMMs (DSPs) on the digital voice daughtercards determines how many channels, which are bidirectional, are available on the card (e.g., 12 bidirectional channels per DIMM; or, a maximum of 60 bidirectional channels with 4 additional DIMMS). The number of simultaneous channels available on a particular digital voice switching daughtercard can be determined easily by counting the DIMMs on the card. The illustration on the next page shows a VSD T1/E1 card fully-populated with standard DSPs and additional DIMMs. As a minimum configuration, three DSPs come standard on the digital cards to provide 12 channels (4 bidirectional channels per DSP); however, it is strongly recommended that only fully-populated (128 MB, 60 channel) VSD daughtercards be installed for the following reasons. · Although a VSD with 12 channels (three std. DSPs) has two operational T1 ports that can provide up to 48 bidirectional channels, only the first 12 simultaneous calls can be handled per DSP; therefore, without additional DSPs (DIMMs) the 13th call and all subsequent calls will be ignored completely, i.e., no dial, busy signal or comfort noise will be generated, until either a channel becomes available, or additional DIMMs are installed. · The installation of additional DIMMs, in effect, provides redundancy in the event of a DSP failure. o Notes o There are no DIMMs on the VSA daughtercard per se, only on the FXO or FXS grand-daughtercards with a maximum of one channel each per port. See Voice Switching Daughtercard -- Analog on page 2-19 for more details. DIMMs are not field upgradeable; however, the flash memory on the boards is field upgradeable. The flash memory must always match the image used or the daughtercard will not function properly if at all. Contact Alcatel's Customer Support for details on obtaining the appropriate flash and/or corresponding image upgrade. Page 2-4 Introduction Top View Digital Voice Port DSP DSP DSP A (4 DIMMs = 48 Channels) D I M M DSP DSP DSP D I M M Switch Bus B Digital Voice Port D I M M DSP DSP DSP D I M M DSP DSP DSP Digital Voice Port Bottom View DSP A DSP 12 Channels Standard DSPs DSP Switch Bus B Digital Voice Port flash Digital Voice Switching Daughtercard (VSD T1/E1) -- DSPs/DIMMS Top and Bottom Views Page 2-5 Voice Switching Daughtercard -- Digital Voice Switching Daughtercard -- Digital The digital Voice Switching Daughtercard (VSD) is used to provide digital telephone connections in Alcatel's H.323 VoIP gateways. There are two main types of digital voice switching daughtercards (VSDs) that can be used to provide VoIP: North American T1 or European E1 (QSIG or Euro ISDN PRI) and (VSBs) Euro BRI ISDN (E1 ETSI). Euro BRI ISDN; see also Voice Switching Daughtercard -- Euro BRI ISDN on page 2-13. Each VSD contains two ports per daughtercard and up to 24 DS0 channels (T1) or 60 DS0 channels (E1) per port. A maximum of one daughtercard can be installed per OA-512 switch (see VSD Front Panel on page 2-7), and up to two daughtercards can be installed in a VSX in each available slot of an Omni Switch/Router. All in all, there are five main daughtercard DSP/DIMM configurations on the VSD version of the digital cards: · 12 channels (0 DIMM; only standard DSPs) · 24 channels (1 DIMMs) · 36 channels (2 DIMMs) · 48 channels (3 DIMMS) · 60 channels (4 DIMMS) Each VSB contains four ports per daughtercard and two ISDN BRI B-channels and one Dchannel per port. A maximum of one daughtercard can be installed per OA 512 switch (see VSB Front Panel on page 2-15), and up to two daughtercards can be installed in a VSX in each slot of an Omni Switch/Router. The VSB supports eight channels via two standard DSPs with four channels each, but does not support any add-on DIMM modules. The OmniAccess 512 chassis provides one empty expansion slot (labeled as S4) reserved for use with features such as Voice Over IP (VoIP); it does not accept the VSX switching module used in Omni Switch/Routers. VSDs, VSBs and VSAs cannot be installed in the same slot in an OSR, and an MPX card is required in the OSR. See VSX Switching Module on page 2-25 for more information. Port numbers can vary depending on the VoIP switch configuration; see also VoIP Daughtercard Port Numbering Schemes on page 2-28. All VSD ports are digital 8-pin, RJ-45 voice ports containing from one to 60 channels per port, depending on whether the voice port interface type is T1 or E1. All VSD and VSB daughtercards require 32 MB Flash memory, and for OSR configurations, 64 MB DRAM memory on the MPX. For power requirements, see VSX Switching Module on page 2-25. The following FCC Class B certifications for the VSD daughtercards have been obtained to date: OA-512-VSD-36T1, -36E1, -48T1, -48E1 and -60E1. o Notes o The number of simultaneous calls per card is dependent upon the number of available DSP channels. Two channels are used per call, e.g., with 12 channels six simultaneous calls can be connected. Calls between channels on the same VSD or VSB card use PCM (Pulse Code Modulation) instead of the H.323 protocol to process digital calls, and require two DSP channels to make the calls. Page 2-6 Voice Switching Daughtercard -- Digital VSD Front Panel Each port has three corresponding LED indicators with link status displays as shown below. Reset Button: Insert pin to reset VSD. (Not available this release.) All VSD daughtercards have three LED displays per voice port as follows: FAIL: On when VSD fails or diagnostic test fails, or when VSD image download fails. Off when VSD hardware is functional, or when VSD image download is OK. VSD A FAIL ERR LINK B A B ERR: On when T1 or E1 VSD voice port port link error occurs in line. This can be any T1 or E1 type of error, e.g., out-offrame/loss of synchronization. LINK: On when T1 or E1 VSD voice port link to switch is connected. Off when signal is lost and T1 or E1 VSD voice port link disconnected. VSD (Voice Switching Digital) T1 or E1 Daughtercard Front Panels Page 2-7 Voice Switching Daughtercard -- Digital VSD Deadman Switch The two types of digital voice switching daughtercards (VSDs and VSBs) contain mechanical relay switches referred to as "Deadman" switches. The Deadman switch is a relay switch that allows two telephony ports using standard telephone connectors, such as the 8-pin RJ-45 jacks used for transmission lines, to be connected to each other in the event of a power failure until 1) power is re-applied to the daughtercard, and 2) the switch reboots to break the Deadman connection and allows VoIP calls to again be placed. For more information on the RJ-45 jacks, see VSD RJ-45 Specifications on page 2-11. The Deadman switch, which resets after 200 ms, also contains a watchdog timer. Because the timer keeps the switch relays open when the Deadman switch disconnects the RJ-45s from each other, they can be connected immediately to the framers terminating the digital or analog telephone line. So, if power is lost to the VSD (or VSB), the Deadman switch keeps the PSTN connection alive by connecting the two telephony ports together before the signals reach the framers. In other words, for all new incoming calls, a connection is maintained between one port connected to a customer's PBX and the second port connected to the PSTN, otherwise known as PSTN fallback. No special configuration is required to use the Deadman switch on the VSD T1 cards; however, to use it on the VSD E1 cards, one port on the card must be set to be the qmaster and the other port must be set to be the qslave. o Note o On VSD E1 cards, setting both voice ports to qmaster (or both ports to qslave) will cause the two telephony switches to get alarms when the Deadman relay switch connects the two ports together. See the digital port configuration commands in Chapter 5, "VoIP Commands," specifically the voice port isdn protocol command used to control the QSIG protocol settings. PBX #1 Digital Voice Port A Deadman Switch PSTN B Normally Open (disconnected) when power is re-applied to card. Switch Bus Digital Voice Port Deadman Switch -- PSTN Fallback Call Protection Page 2-8 Voice Switching Daughtercard -- Digital VSD Cross-Over Toggle Switch The Cross-Over toggle switch for digital voice daughtercards (VSDs only) can be used to correct communication link errors between a daughtercard in the switch and a PBX or keyset due to the transmit (TX) and receive (RX) pins of the cable connecting the VoIP daughtercard and the digital telephony device (PBX or other voice device). This will show up as the link LED not turning green (see VSD Front Panel on page 2-7). If a communication link error occurs between the switch and the PBX or Key Set as such, the blue Cross-Over toggle switch, as shown here on the top side of the board, can be flipped to a Cross-Over ON or OFF position after shutting down the VoIP switch and removing the affected daughtercard. This will swap the transmit and receive connections for the designated port. The default toggle position is to the left or OFF position. Once the toggle switch has been flipped, the card can be reinstalled, and the ports on the voice daughtercard reconnected to the PBX or other voice device using either a StraightThrough or Cross-Over cable (Straight-Through recommended). o Notes o An amber cellophane tape may need to be peeled off the top of the Cross-Over toggle switch before the toggle switch can be flipped. When the tape is present it indicates the toggle switch is set to factory default. The physical port always has 8 pins, but changes functionally depending on the cable in use. For more information on the RJ-45 jacks, see VSD RJ-45 Specifications on page 2-11. Straight-Through Default Digital Setting Voice Port to Left Top View DSP DSP DSP D I M M DSP DSP DSP D I M M A Cross-Over Switches (T1) Switch Bus DSP D I M M DSP D I M M B Digital Voice Port DSP DSP DSP DSP Cross-Over Switches -- Swapping Port Transmit/Receive Connections Page 2-9 Voice Switching Daughtercard -- Digital Cabling Of the common cable types (compatible with RJ-45 jacks) that can be used with VoIP switches, the Straight-Through (Ethernet) and Straight-Through (T1 Voice) are both acceptable, as well as the Cross-Over (T1) cable; however, due to the TX/RX pinout wiring, the Cross-Over (Ethernet) cable cannot be used with VoIP switches. If the Cross-Over (Ethernet) cable is used the LINK LED will not display. o Note o For E1 configurations, it is recommended that a balun connector always be used to connect a voice device (e.g., PBX) that uses an ITU G.703 interface (coaxial cables, BNC connectors) to a VSD (or any RJ-45 E1 port). The balun converts the impedance of 120 Ohms on the RJ-45 port to 75 Ohms (G.723). The balun connector is not required when both ends have RJ-45 connections. Contact Alcatel's customer support for more details on balun connectors. Page 2-10 Voice Switching Daughtercard -- Digital VSD Pinouts The following illustration shows the pinouts for the digital voice switching daughtercard (VSD) 8-pin, RJ-45 jacks used to connect the voice ports on the card to voice devices in the VoIP network that support digital connections, e.g., PBX and Key Set. o Note o The pinouts as shown indicate when the Cross-Over toggle switch is ON and OFF. VSD RJ-45 Specifications Pin Number 1 1 8 Standard Signal Name Receive Data + Receive Data - 2 3 4 5 Transmit Data + Transmit Data - OFF Cross-Over Toggle Switch 6 7 8 VSD RJ-45 Specifications Pin Number 1 1 8 Standard Signal Name Transmit Data + Transmit Data- 2 3 4 5 Receive Data + Receive Data - ON Cross-Over Toggle Switch 6 7 8 Page 2-11 Voice Switching Daughtercard -- Digital VSD Jumpers The following jumpers are factory set on the VSD daughtercard and should not be changed by the customer unless under the direction of Customer Support. Note that, in general, only jumpers which can be set with shunts, or are associated with ports on the board are identified and described. o Caution o This information is being provided solely for the purpose of repositioning a shunt which may have been inadvertently removed, so as to prevent damage to the board, and/or possibly render the board or other components connected to the VSD inoperable. Jumper No. P3 P3 P7 P7 P7 P7 P10 P10 Shunt Position no shunt (on Pins 1_2) no shunt (on Pins 3_4) no shunt (on Pins 1_2) no shunt (on Pins 3_4) no shunt (on Pins 5_6) no shunt (on Pins 7_8) Pins 1_2 Pins 3_4 Default yes yes yes yes yes yes yes yes Port B A B B A A B A Description RJ-45 connection RJ-45 connection RJ-45 connection RJ-45 connection RJ-45 connection RJ-45 connection RJ-45 connection RJ-45 connection Other VSD Jumpers Jumper No. P17 Shunt Position Pins 2_3 Default yes Description Backplane interface (Hbus) Page 2-12 Voice Switching Daughtercard -- Euro BRI ISDN Voice Switching Daughtercard -- Euro BRI ISDN The Euro BRI voice switching daughtercard (VSB) is used to provide the European ISDN BRI voice port connections in Alcatel's H.323 VoIP gateways. Because many aspects of this card are similar to the VSD T1/E1 previously discussed; see also VoIP Daughtercard Types on page 2-2 and Voice Switching Daughtercard -- Digital on page 2-6. Digital Signal Processors (DSPs) and Available Channels Unlike the VSD T1/E1 VoIP daughtercard shown previously, the VSB card does not support any additional DIMMs, only the two standard DSP which provide the VSB with eight channels. See also Digital Signal Processors (DSPs), DIMMs and Available Channels on page 2-4 for a more details. VSB Deadman Switch There are two Deadman switches on the VSB daughtercards. One Deadman switch is for ports A and B (1 and 2) and the other is for ports C and D (3 and 4); on a VSX switching module with two VSBs, the relays switches on the second card would be for ports A and B (5 and 6) and for ports C and D (ports 7 and 8). For more details on the Deadman switch, see VSD Deadman Switch on page 2-8. Port numbers can vary depending on the VoIP switch configuration; see also VoIP Daughtercard Port Numbering Schemes on page 2-28. o Note o For the deadman switch to operate properly on the VSB, ports 1 and 3 must be configured as TE, and ports 2 and 4 must be configured as NT (ports are TE, NT, TE, NT). VSB NT (LT)/TE Cross-Over Toggle Switch There are four NT (LT) / TE Cross-Over toggle switches on the VSB daughtercards. Each switch is factory set to NT. For more information on ISDN terminators, seeVSB Jumpers on page 2-16, and also Chapter 3, "Network Dialing Schemes." The NT (LT)/TE toggle switch is physically similar to the VSD Cross-Over toggle switch on the VSB, but functionally different as it is used to select either a network terminator or terminal equipment as an endpoint. For a description of the Cross-Over toggle switch on the digital voice switching daughtercards, see VSD Cross-Over Toggle Switch on page 2-9, and Cabling on page 2-10. NT (LT) means that each NT port emulates the "network" side or "line terminator" point of the ISDN connections to the ISDN/PSTN network, e.g., connections to PBX, Key Set, BRI TelSet, Group 4 (ISDN) facsimile machine. TE means that each TE port emulates the "terminal" side of the ISDN connections to the ISDN network, e.g., PBX, Key Set, CO (Central Office) switch and ISDN telephone switch. VSB Pinouts The Euro BRI VoIP daughtercard (VSB) uses the 8-pin, RJ-45 jacks in the figures and tables on the following page to connect the voice ports on the card to voice devices in the VoIP network that support digital connections, e.g., PBX, Key Set, BRI TelSets, Group 4 (ISDN) facsimile machine, CO (Central Office) switch, and ISDN telephone switch. For more details on the VSB pinouts, see VSD Pinouts on page 2-11. Page 2-13 Voice Switching Daughtercard -- Euro BRI ISDN VSB Configured as TE RJ-45 Specifications Pin Number 1 1 8 Standard Signal Name Unused Unused Tx + Rx + Rx Tx Unused 2 3 4 5 6 7 8 VSB Configured as NT RJ-45 Specifications Pin Number 1