Asterisk & VoIP and it’s role in your enterprise

Asterisk? Open-source software released under the GPL Sponsored by Digium, the main hardware provider for POTS interface cards Digium named in the top 10 open source companies to watch by networkworld.com Ports for most *nix systems including Solaris Ports also available for OSX and Windows Open standards along with some proprietary protocol support (like Cisco’s Skinny and MGCP) Modular plugin type system

Protocols Supported SIP – Session Initiation Protocol H.323 – Common in video conferencing Skinny – Cisco IP Phones default protocol MGCP – Media Gateway Control Protocol IAX - Inter-Asterisk Exchange Protocol Codecs Supported G.711 – Best voice quality ~100Kbps G.729 – Good voice quality ~40Kbps GSM – Acceptable voice quality ~10Kbps G.722, G.723.1, G.726, iLBC, Linear, LCP-10, Speex

PSTN Interface Support Analog FXO FXO FXS FXS E&M (w/ or w/o Wink), Loop start, Ground start, Kewl start E&M (w/ or w/o Wink), Loop start, Ground start, Kewl startT1 E&M (w/ or w/o Wink) E&M (w/ or w/o Wink) Robbed bit Robbed bit ISDN (PRI & BRI) 4ESS 4ESS Lucent Lucent National National Some international support Some international support

Biggest Features Unified Voic Voic to (.wav files) Voic to (.wav files) Video Conferencing messages to (.mpg/.avi files) Video Conferencing messages to (.mpg/.avi files) Advanced Meetme conferencing - conference bridging Web management interface Web management interface Support for conference numbers and passwords Support for conference numbers and passwords Presenter and Presentee support (presenter can mute all participants, etc) Presenter and Presentee support (presenter can mute all participants, etc) Contact Center Queuing Interactive Voice Response Automated Attendant Video Conferencing (SIP and H.323) Jabber / Google Talk integration Find me / Follow me Out of state DIDs (all VoIP systems) Call monitoring and recording

Normal PBX Features Name it, it’s there

Enterprise Features Unified dialplan across many servers SQL Compliant databases (through ODBC) SQL Compliant databases (through ODBC) Native support for Mysql Native support for Mysql LDAP integration LDAP integration DNS SRV records Make calling each other easy Make calling each other easy Scalability Scalability Load balancing Load balancing Phone provisioning Phone provisioning DUNDI – Distributed Universal Number Discovery Works with SIP proxys to accept large amounts of phone registrations AGI – Extend your system using C, C++, Perl, PHP, … Custom CDR – Used for calling card integration, billing, …

Asterisk Compatible Endpoints Cisco IP Phones (except IP 7920) Polycom IP Phones Snom IP Phones Avaya IP Phones Linksys IP Phones Many others Asterisk Compatible Gateways Cisco VoIP Gateways (anything MGCP, SIP, H.323) Cisco Callmanager (through SIP and H.323) Patton Smartnode

Asterisk PSTN Interfaces Digium Analog (up to 24 channels on a single PCI card, FXO and FXS) with hardware echo cancellation Analog (up to 24 channels on a single PCI card, FXO and FXS) with hardware echo cancellation T1 / PRI / BRI – 1-4 on a single PCI card with hardware echo cancellation T1 / PRI / BRI – 1-4 on a single PCI card with hardware echo cancellationSangoma Better analog support, but uses more PCI slots (or spaces) Better analog support, but uses more PCI slots (or spaces) T1 / PRI / BRI – 1-8 on a single PCI card with hardware echo cancellation T1 / PRI / BRI – 1-8 on a single PCI card with hardware echo cancellation Clear channel DS3 Clear channel DS3 Any SIP / H.323 compliant endpoint Cisco x8xx series ISR routers Cisco x8xx series ISR routers …

Possible Asterisk Configurations Full PBX Add small remote sites onto traditional PBX Trunk Routing Gateway Toll Bypass Only Service Component

Full PBX

Traditional PBX with Remote Offices Use of Digium appliance Cisco x8xx routers

Trunk Routing Gateway Connect a traditional PBX to services like Sprint SIP Trunking

Toll Bypass Only Connect Traditional PBXs together over WAN Links

Service Component Meetme Conferencing Bridge Voic / Unified Messaging Add Softphones to non-VoIP PBX …

Drawbacks No PCI-Express Support (just came out for digital interfaces, still none for analog interfaces) Hardware sizing information hard to find Kernel updates break Digium drivers PCI Bus sharing can cause significant problems with voice quality NAT Traversal (common across any SIP system) Linux system QoS not very mature yet No VoIP security yet, although planned Not for *nix beginners

Support Certifications – dCAP (Digium Certified Asterisk Professional) Voip-info.org (VoIP WIKI site) Commercial support (through Digium) Partners (certified or un-certified) Mailing Lists (very active) User Groups (although none in Boston) Astricon (Asterisk conference) Books

Where to start? (Asterisk main page) (Asterisk and CentOS with full installer) (huge resource for VoIP related projects and configuration info) (hardware manufacturer and project sponsor) Trixbox (formerly HOME) User Groups – Find one close or we can try to start one Books – O’Reilly, Asterisk for Dummies, Asterisk configuration guide, …