VoIP Series - Build, Test, and Deploy VoIP Applications with Asterisk and other Open-Source Applications

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1 VoIP Series - Build, Test, and Deploy VoIP Applications with Asterisk and other Open-Source Applications Elliot Eichen Tue Jan 29, 01-02:30pm, No enrollment limit, no advance sign up This session will provide an overview of the open-source toolbox for Voice over IP (IP-PBXs, SIP Proxies and User Agents, Protocol and Media debugging, Codecs, Signaling and Media Testing, etc.). Particular emphasis will be placed on using Asterisk (distributions, architecture, features, configuration, and applications). The session will include discussion and demonstration. Contact: Elliot Eichen, N42-169, x3-8647,

2 Outline 1. The mostly open-source toolbox: IP-PBX SIP Proxy SLEE (service layer execution environment) Stacks Protocol Analysis Network emulator Voice Quality Testing Load & Signal Generator Network Providers Codecs AAA (accounting, authentication, authorization) Hacking/Eavesdropping 2. Asterisk Fundamentals: Distributions Definition & Functionality API: definition and examples 3. Asterisk Demonstrations: Asterisk config utility Command Line Interface ngrep protocol analysis & ladder diagram Sample programs

3 The mostly open-source IP Telephony toolkit + IP-PBX: Asterisk The mother open-source IP Telephony app. A veritible industry onto itself, many distributions, many protocols (H323, MGCP, SIP, IAX, etc.). Commercial support available, used by enterprises, ITSPs, ASPs, etc. SIP Proxies: IPTel Sip Express Router. Commercial version available SER (TEKELEC), used by ITSPs, some platforms. openser Romainian branch of SER. Used by used by Columbia/UPenn deployments, MIT Personal SIP, other apps. JAIN SLEE Java Service Layer Execution Environment. Decendent (at least in spirit) from the US Nat'l Inst. of Standards SLEE (still available), reference implementation supported by Open Cloud (New Zeeland), support from SUN?. +Note: these choices are purely personal, based on my own interests, what was available at the time that I was looking, references from friends, or perhaps just whimsy. There is no intention of endorsing one application over another. Browse through voip-info.org, or sourceforge, the number of applications is astounding!

4 IP Telephony Toolkit: SIP User Agents & Stacks User-Agents: xlite minisip Free, not open-source, but very commonly used softphone. Windows and Mac. Older versions include SIP messaging, removed in the later versions. client with SRTP, reported avaialble for WinCE, Royal Institute of Stockholm Stacks: For historical purposes, supported by Cisco, book "Practical Vovida VoIP." Java, don't think any development in several years. Founded and supported by Pingtel (early Java based IP Phone SIP Foundary startup, Boston). resiprocate and SIPX seem to both point back to SIP Foundary. Sofie Nokia's SIP stack, seems to reasonably active.

5 IP Telephony Toolkit - Testing Protocol Analyzier: Foremerly "Ethereal." Mother of open-source protocol Wireshark analyziers, builds ladder diagrams and provides RTP playback. "network grep", lightweight, very handy for looking at ngrep signaling events remotely Network Emulator: NISTNET Simulate delay, jitter, packet loss, various models. Still useful, although development stopped. Voice Quality: Patented, reference code avaialble to ITU members for ITU Code PESQ standard. Several commercial implementations Load & Signaling (protocol) Testing Useful Q/A tool, regression test under load SIP network SIPP elements. Some ability to test media. Supported by HP.

6 IP Telephony Toolkit: Carriers & Miscellaneous Apps Network Providers: One of many, free on-net, SIP carriers straightforward, FreeWoldDialup some useful tools (loopback). Have been using them as my IXC for many years. Voicepulse Reasonable prices for DIDs and PSTN termination. Some complain about quality. Interesting for peer-peer & business models, & the Skype fact that they actually made $$ doing VoIP. A number of SIP<=>Skype gateways, no personal exerpience. Other Stuff: SIPit (used to be called SIP Backoffs, but Pillsbury SIP Forum complained). Interop orgies. Free cracking tool, provides a man-in-the-middle Cain & Able attack (ARP Table Posioning) to intercept calls. AAA, CDRs, etc. Everyone seems to use this, not sure Free Radius about if/when Diameter (IMS) will be supported.

7 IP Telephony Toolkit: Open-source CODECS Native PSTN codec, pretty much the reference standard. Just PCM (8k G711 samples/sec x 8 bits/sample = 64 kb/s) with compounding. Some confusion around whether the original 13 kb/s GSM codec GSM (developed for cell phones) is patented, there is code available, folks have treated this as being open-source. VoIP MGW support? Developed by GIPS (Global IP Sound, owned a piece of Skype), it's 15.2 kb/s in 20 msec frames, supposed to have better voice quality and ilbc higher resiliency to packet loss than G279a. Free, source code available, not open source. Cisco MGW support. Open-source compressed codec project, supposed to get around all the Speex patent issues. Interesting for their wideband (32KHz, 44 kb/s stereo) capability. No MGW support. H264/ MPG4 Reference implementation available from the ITU, commercial implementations patented, see the MPEG Licensing Authority. Note: Lots of drama (patent issues) around codecs, clearly a hot area for delivery of music and video as well as voice.

8 Asterisk Distributions Digium: Build Asterisk directly from Digium CVS. Time consuming (at least for me, due to all the libraries and dependencies), but probably the fastest way to get the latest code. Trixbox: Australian asterisk camp, FreePBX is the configuration utility, also an operator console and visual voic . Some people complain that it bundles too much stuff (some of it not so good). This is what I have been using Digium s answer to Trixbox, I have not tried this. Astfin: embedded on Blackfin running uclinux. Free Telephony Project provides reference platform & sells a box, other commercial boxes based on Blackfin DSP (Analog Devices). Key system replacement, low electrical power, very interesting. OpenWRT: poor man s embedded linux hardware (e.g., Linksys WRT54G, others). Distributions with either Asterisk or SER. Limited horsepower, not good for media things (like echo cancellation, transcoding between codecs, perhaps voic , etc.).

9 What is an IP-PBX Typically, SIP Registrar & Proxy functionality provided by a back-toback user agent (proxy functionality) + feature server + voic . IP Centrex platforms (often called Hosted IP-PBXs by carriers who want the PBX business) => Multi-tenant IP-PBX Perhaps some religious differences between IP-PBX (e.g asterisk) disciples and Proxy (e.g., openser) disciples around proxy vs. back-to-back UA functionality. B:B User Agent model Proxy Model

10 Long Term MITvoip Architecture 5/9/2007 ~ 1-2,000 Open Source? ~ 13,000 IP-Centrex ~ 2,000 Analog Telephone Lines Emergency Phones Fax Session Border Controller (SBC) Modem, point of sale, alarms, etc. Internal Access (Line Side) Location db Outbound Proxy Pilot:openSER SYL Presence Server SYL Syl Control Presence Server (feature Server server, registration db) PBX.. Shared Services ENUM (DNS) Internal (routing) Proxy Pliot: openser SYL Route Server SYL Route Server Messaging (Iperia) Other Shared Services (conferencing, presence, etc.) External Access (Trunk Side) DMZ SBC Carrier SBC VoIP Gateways Pilot:Cisco 5850 New VoIP Platform Pilot Platform New Analog Platform Public IP PSTN

11 Asterisk API Asterisk Application Gateway Basically, the API allows you to pass execution to UNIX process. You can write/read from/to Asterisk using STDIN, STDOUT, and you can see what s going on via the Asterisk console (CLI) by using STDERR. {Note talk re: forking} Invoking the process looks like standard Asterisk handling of a call: exten => 3050, 1, Answer exten => 3050, n, AGI(foo.agi arg1 arg2 arg3); std asterisk exten => 3050, n Asterisk also passes a bunch of internal variables to foo (without asking). foo now has control of the call, and can do programmatic things (write to web pages, send SMS messages, read from databases, etc.), and execute standard Asterisk primitives (play streams, collect digits, transfer, etc.) In addition to AGI (which is really control over the how the call gets executed), there are three related application calls: EAGI: control the sound channel in addition to interaction with the dial plan. FastAGI: run the application a remote machine. DeadAGI: provides access to a dead channel, after hang up.

12 Live Demonstration(s) 1. Asterisk config utility Trunks, Extensions, Inbound IVRs, etc. Asterisk file structure extensions.conf, sip.conf, etc. 2. Command Line Interface Debugs, endpoints, etc. 3. Protocol analysis & ladder diagram Client, ngrep, wireshark 4. Sample programs test-agi.agi extension 3050 (simple) Conference Bridge (more complicated)

13 Example: extensions_additional.conf:exten => 3050,1,Answer extensions_additional.conf:exten => 3050,n,AGI(agi-test.agi) extensions_additional.conf:exten => 3050,n,Hangup == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/ cde718' -- Executing Answer("SIP/ c60390", "") in new stack -- Executing AGI("SIP/ c60390", "agi-test.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-test.agi -- Playing 'beep' (escape_digits=) (sample_offset 0) -- <SIP/ c60390> Playing 'digits/1' (language 'en') -- <SIP/ c60390> Playing 'digits/hundred' (language 'en') -- <SIP/ c60390> Playing 'digits/90' (language 'en'). -- <SIP/ c60390> Playing 'digits/60' (language 'en') -- <SIP/ c60390> Playing 'digits/5' (language 'en') -- Playing 'testagi' (escape_digits=) (sample_offset 0) -- AGI Script agi-test.agi completed, returning 0 -- Executing Hangup("SIP/ c60390", "") in new stack

14 Appendix: Asterisk Primitives answer Answer channel receive char Receives one character from channels supporting it channel status Returns status of the connected channel receive text Receives text from channels supporting it database del Removes database key/value record file Records to a given file database deltree Removes database keytree/value say alpha Says a given character string database get Gets database value say digits Says a given digit string database put Adds/updates database value say number Says a given number exec Executes a given Application say phonetic Says a given character string with phonetics get data Prompts for DTMF on a channel say date Says a given date get full variable Evaluates a channel expression say time Says a given time get option Stream file, prompt for DTMF, with timeout say datetime Says a given time as specfied by the format given get variable Gets a channel variable send image Sends images to channels supporting it hangup Hangup the current channel send text Sends text to channels supporting it noop Does nothing set autohangup Autohangup channel in some time receive char Receives one character from channels supporting it set callerid Sets callerid for the current channel receive text Receives text from channels supporting it set context Sets channel context record file Records to a given file set extension Changes channel extension say alpha Says a given character string set music Enable/Disable Music on hold generator say digits Says a given digit string set priority Set channel dialplan priority say number Says a given number set variable Sets a channel variable get full variable Evaluates a channel expression stream file Sends audio file on channel get option Stream file, prompt for DTMF, with timeout control stream file Sends audio file on channel and allows the listner to control the stream get variable Gets a channel variable tdd mode Toggles TDD mode (for the deaf) hangup Hangup the current channel verbose Logs a message to the asterisk verbose log noop Does nothing wait for digit Waits for a digit to be pressed

15 Example agi script:

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