VXML Solutions

What is SIP and why do I care?

Tropo sits in Voxeo’s live communications cloud; an environment specifically designed for real-time voice, instant messaging and SMS applications. On the voice side, a key feature of the Tropo platform is its extensive support for SIP. The Session Initiation Protocol is a Voice over IP (VoIP) technology enabling two ends of a conversation (or session) to negotiate media formats much like the Accepts/Content-Type header does for HTTP.

As a voice provider, SIP is the best way to guarantee seamless interoperability between existing phone systems, carriers and call centers. Not having SIP is like playing Call of Duty on a 15” with no HDMI. Sure it technically works…. but why?!?

Read on for real world example.

In this example, Acme Flowers has an Asterisk based phone system that rings two sales people (Bob and Alice) when a call comes in on 555-1212. Before integrating with Tropo, the phone system would simply drop the caller into voicemail if no one was available.

Tropo FTW! With a couple tweaks, callers are now automatically transferred to a voice-driven menu in the Tropo cloud, providing them with driving directions, order status and store hours all over the Internet, all with industry leading speech synthesis and speech recognition. No extra phone line. No additional telco charges. On top of that, customers get all the same services via SMS and IM using the same exact code!

SIP allows Tropo applications to seamlessly integrate with existing phone systems and call centers over the Internet; cutting the telco out entirely!

Enabled on every app!

Tropo assigns a personal SIP address for every application. This special address is accessible from any modern phone system or SIP-enabled device. You can also use  a soft phone like X-Lite for Mac/PC or iSIP / fring for the iPhone.

Give it a try now by calling my demo app at sip:9991429776@sip.tropo.com

SIP Headers

Ever been asked for your account information 20 times only to be transferred to a live person who has no idea who you are or why you’re calling? The problem lies in that most phone systems are a black box, blindly sending calls from one department to another. SIP solves this problem by allowing phone systems to exchange information in between hops. 

SIP borrows many concepts from HTTP; one of which is the concept of a header. When SIP endpoints communicate, they send key/value pairs that help route the call. Wouldn’t it be nice if you could leverage this pipeline to communicate non-telephony data? With Tropo, you certainly can. 

Tropo exposes SIP headers to your application on every incoming call allowing your application to easily integrate with other systems.

Further reading

To learn more about SIP and related technologies check out these sites:

• SIP Overview @ Wikipedia – http://en.wikipedia.org/wiki/Session_Initiation_Protocol
• VoIP Info – http://www.voip-info.org/wiki/view/SIP
• IETF SIP Specification (not for the faint of heart) – http://www.ietf.org/rfc/rfc3261.txt

That’s it for now. I hope to follow up this post with more examples of using Tropo and SIP to unlock you calls.

Until then,

Good Hunting

Originally from Voxeo Blogs

The W3C Voice Browser Working Group recently released the fifth working draft of the VoiceXML 3.0 specification at:

http://www.w3.org/TR/2010/WD-voicexml30-20100304/

This newest draft made the following changes:

• Added V2 Convenience Syntax appendix, referenced from convenience syntax section..
• Updated Core concepts (sections 4.1 and 4.2)
• Added section stub for the Subdialog Module
• Added Disconnect Module
• Added Connection Resource
• Expanded the description of the Basic Profile
• Added placeholder for new “Enhanced” profile
• Added Play RC
• Added SCXML code and issue list to Grammar module
• Added Record Module
• Updated language identifier definition and link to BCP47
• Added Property Module
• Added basic Transition Controllers text to the Document Initialization and Execution section

A diff between this draft and the previous draft is available at:

http://www.w3.org/TR/2010/WD-voicexml30-20100304/diff.html

Given that Voxeo’s Dan Burnett is the co-Editor-in-Chief of the VoiceXML 3.0 specification, you can expect to see more postings here about VoiceXML 3.0 as it continues to move along the path toward becoming an actual standard.

Originally from Voxeo Blogs

2010 will be held this Wednesday March 10, 2010 at the Sheraton Safari Hotel and Suites in sunny Orlando, Florida (where Voxeo headquarters is also located).  Splunk lets you search and analyze all your IT infrastructure data from one place in real time. With Splunk, you can troubleshoot application outages, investigate security incidents and more in minutes, not hours or days.  Voxeo’s CTO, RJ Auburn will be doing a customer presentation at this event and you won’t want to miss it!

Originally from Voxeo Blogs

A hybrid IVR platform is one that is available as a hosted/cloud-based solution and as an on-premise deployment. Just to be clear, we are not simply talking about two distinct offerings delivered by a single vendor. We mean one platform – offered two ways.   Benefits include the ability to migrate between deployment models at any time or to take advantage of both deployment models at the same time — with no changes whatsoever to your applications. That’s exactly what our partner Tele-Works did to enable Harris Computer Systems to serve a diverse base of utility customers.

Harris Computer Systems provides Customer Information Systems, billing software and other solutions to thousands of utility customers throughout North America. Harris recognized that IVR is a key component to a successful utility’s customer service department, but required a way to serve all of their customers – from the largest to the smallest – with one consistent solution.

Tele-Works provided a Voxeo-powered solution that uniquely supports both hosted and on-premise deployments, as well as inbound and outbound calling, from a single code base. The Tele-Works solution extends across Harris’ complete utility product portfolio, giving thousands of utility customers access to quality IVR at price points that were previously unattainable.  Read the case study here.

Originally from Voxeo Blogs

The final VXI* VoiceXML browser 4.4  ref. 2010-03-05 32bit and 64bit is now released. This new release is suitable for production platforms running with all lastest Asterisk 1.4 and 1.6 kernels (both Asterisk’s packages are available for download too). Like our previous release, VXI* 4.4 has been built to run over Asterisk EC2 or Xen virtual servers. Stay tuned to the blog to keep up to date on our progress or check out our lasts builds.

You can download these new binary packages from this website for registered users.

» Linux 32bit : Debian Etch | Debian Lenny | CentOs 5 | Debian Sarge | More…
» Linux 64bit : Debian Etch | Debian Lenny | CentOs 5 | More…

Powered by I6NET Software

New features added and modifications:add: Complete DTMF buffering during HTTP long requests.

• add: Add paramter threshold to configure the VAD/silence (record).
• add: Add parameter autoexit to kill asterisk if the connection with VXI is lost.
• add: Set record maxtime shadow variable.
• add: Improve prompt hangup and bargein (skip HTTP processing, limit queue-fill).
• mod: Select the first account with redirection(s).
• add: Add clean support of noinput and hangup event during the record.
• add: Add the account parameter “force” to set Transfercapability=VIDEO.
• mod: Improvement of the bridge transfer (use with transcode).
• mod: Disable the msgqlock.
• add: Add parameter videoprofile (to controle the video codec transcoder).
• add: Check the account in the vxml(@) execution.
• mod: Correction to control the call answer.
• add: bridge and spawn modes for localformat.
• mod: Add the DOCTYPE in the grammars.
• mod: Correction in the session release (wait for playall).
• mod: Correction for better speech support.

Originally from i6net

Text to speech engines have long been used to allow those who cannot talk to communicate verbally. Film critic Roger Ebert, who lost his lower jaw and his voice to cancer, has taken it a step further by creating a TTS voice that sounds like him.

Using the hundreds of hours of archived film clips from his reviews and other TV appearances, Ebert’s voice was reconstructed by Scotland’s CereProc, a developer of text to speech technology.

Debuting his new voice on Tuesday on Oprah, Ebert said, “You’ll know it’s a computer, but one that sounds like me. It still needs improvement but at least it sounds like me. In first grade they said I talked too much, and now I still can.”

Originally from Voxeo Blogs

Getting noticed in the job market today gives you an edge. One edge is to create an audio resume that you may give out to show your developer prowess. Using the Tropo multi-channel capabilities, you may then allow that resume to interact with your perspective employer via SMS or even Instant Message.

Whats more, using Tropo Scripting you may create this resume and host it on our servers with no need for your own server or sending mark-up back and forth. And, you may do this in Groovy, Javascript, PHP, Python, Ruby or all of them to show off your polyglot skills.

For this example resume we will use a Ruby script hosted on Tropo.com to make John Doe’s background available (also available on Github here):

answer
say "Welcome to John Doe's audio resume."

listening = true
while listening
  result = ask "Please press 1 to hear an overview of John's resume, please press 2 to listen to his most recent experience or press 3 to connect to John and hire him!", { 'attempts' => 3, 'choices' => '1, 2, 3', 'onBadChoice' => lambda { say 'Invalid entry, please try again.' } }

  case result.value
  when '1'
    say 'Adept at managing all aspects of information technology, including ongoing business needs assessment, software development and implementation.'
    sleep 2
  when '2'
    say 'John served as the IT Director at ABC Insurance. John is a direct manager of a team of 6 people charged with delivery of enterprise level projects.'
    sleep 2
  when '3'
    transfer 'tel:+14155551234'
  else
    say 'Thank you for your time, goodbye.'
    listening = false
  end
end

hangup

You may call and listen to John Doe’s on 1 (408) 940-5947. To get started writing your own you may register for a free Tropo developer account, with a free phone number, here.

Originally from Voxeo Blogs

Want to learn about the Session Initiation Protocol (SIP)? Would you like to understand how the SIP protocol works and why it is the dominant open standard for communication today? Want to understand the challenges SIP faces and what’s being done to overcome them?

If so… and if you will be attending VoiceCon in Orlando, FL, March 22-25, you’ll be able to join my (Dan York) 3-hour tutorial on “SIP Fundamentals and Prospects” on Tuesday, March 23rd, from 2-5pm. The abstract VoiceCon has posted is this:

SIP (Session Initiation Protocol) has become the dominant protocol for IP communications. This workshop explains SIP — how it works, the major issues impacting deployments and how SIP will evolve in the future.

The session focuses on the technical aspects of SIP and how it is used. It analyzes in detail the major components of SIP architecture, SIP addressing and registration, session establishment, SIP message routing and connecting SIP across the PSTN. You will learn about SIP extensions and how SIMPLE works for IM/presence. The workshop also examines some of the challenges SIP faces, including NAT traversal (and the tools developed to cope with it: STUN, TURN and ICE) and security. The tutorial concludes with an assessment of how SIP may evolve and its role in peer-to-peer environments. You will receive an inventory of SIP resources—books, papers and organizations.

I’m very much looking forward to the session… although I still do have some work to finish up on the materials. For the past while my friend David Bryan has given these tutorials at VoiceCon events, but given that he also chairs IETF working groups he would need to clone himself since this VoiceCon is the same week as IETF 77 in Anaheim, California. It’s a wee bit hard to flip between coasts… and as anyone who has ever been to an IETF event knows, the meetings are intense and he is needed out there.

If you can’t attend VoiceCon this year, I’ll probably do some SIP tutorial webinars in the future and perhaps you’ll see something popping up over at Voxeo University… stay tuned. And if you are at VoiceCon, please do stop by and say hello… or send me an email in advance letting me know.

Originally from Voxeo Blogs

Find all training-related information of Voxeo University on our new website:

• Training Catalog for Prophecy and VoiceObjects courses
• Training Calendars 2010 for Training Centers in Orlando and Cologne – incl. dates of our FREE trainings
• Registration Form incl. pricing information
• Online registration tool

Please help yourself. For other questions or feedback please mail to university@voxeo.com.

Originally from Voxeo Blogs

FYI, a new version -08 of my P-Charge-Info Internet-Draft is now available:

http://www.ietf.org/id/draft-york-sipping-p-charge-info-08.txt

For an understanding of what P-Charge-Info is all about, read why I first wrote it, P-Charge-Info and incredible disconnect between PSTN billing and the new world of SIP, and then my update last year on the -07 draft.

Version -08 really only has a minor tweak to the ABNF notation for the “npi-value” and then a new Appendix A clarifying the npi-values and their relation to ANSI T1.113.

I am hoping that I can very shortly request IESG consideration to move this document along the path to being an RFC. The only remaining issue is that my co-author, Tolga Asveren, has brought forward a proposal for simplifying the parameters a bit. I’ve forwarded that proposal to several people I know are very interested in this draft. We’ll see where it goes from there. I’d very much like to move this along soon, so we’ll see.

Originally from Voxeo Blogs