1 / 12

Integrating VoiceXML with SIP services

Integrating VoiceXML with SIP services. Kundan Singh, Ajay Nambi and Henning Schulzrinne Columbia University {kns10,an2029,hgs}@cs.columbia.edu. What is VoiceXML?. A language for specifying voice dialogs in interactive voice response systems Information retrieval

moisesk
Download Presentation

Integrating VoiceXML with SIP services

An Image/Link below is provided (as is) to download presentation Download Policy: Content on the Website is provided to you AS IS for your information and personal use and may not be sold / licensed / shared on other websites without getting consent from its author. Content is provided to you AS IS for your information and personal use only. Download presentation by click this link. While downloading, if for some reason you are not able to download a presentation, the publisher may have deleted the file from their server. During download, if you can't get a presentation, the file might be deleted by the publisher.

E N D

Presentation Transcript

  1. Integrating VoiceXML with SIP services Kundan Singh, Ajay Nambi and Henning Schulzrinne Columbia University {kns10,an2029,hgs}@cs.columbia.edu

  2. What is VoiceXML? A language for specifying voice dialogs in interactive voice response systems • Information retrieval • News, sports, traffic, stock quotes, voice-mail • e-business • Customer service, banking, stock trading • Notification service SIP/VoiceXML @ Columbia University

  3. Traditional IVR • Receives incoming PSTN5 call • Responds back with prompts • Accepts user input (DTMF or speech) • Takes action based on user input • (Usually the service logic is programmed for the specific application, say weather report) PSTN End user Welcome to voice mail. Press 3 to listen to new messages... 1-212-8545224 IVR1 platform • Voice and telephony functions • (ASR2, TTS3, DTMF4) • Service logic (application specific) [1] Interactive voice response [2] Automated speech recognition [3] Text to speech [4] Dual tone multi-frequency (touch tone) [5] Public switched telephone network SIP/VoiceXML @ Columbia University

  4. Decomposition Internet End user Voice gateway • Voice and telephony functions Web server • Service logic PSTN End user IVR platform • Voice and telephony functions • (ASR, TTS, DTMF) • Service logic (application specific) SIP/VoiceXML @ Columbia University

  5. VoiceXML Multimedia End user Scripts DB Audio/ grammar Web server PSTN Internet End user VXML Voice gateway HTML • Voice and telephony functions • VoiceXML browser Web server • Service logic (CGI, servlet, JSP) SIP/VoiceXML @ Columbia University

  6. HTML vs VoiceXML <form> <field name=‘id’> <prompt> Your ID, please. </prompt> </field> <block> <submit next=“url”/> </block> </form> <form action=“url”> Enter your Id: <input name=‘id’> <input type=‘submit’> </form> Telephony, speech Synthesis or audio output, user input and grammar, program flow, variable and properties, error handling, … SIP/VoiceXML @ Columbia University

  7. Further decomposition End user PSTN Internet End user Voice gateway Voice and telephony function VoiceXML browser Web server • Service logic (CGI, servlet, JSP) SIP/VoiceXML @ Columbia University

  8. Internet telephony Media server (RTSPd) SIP softphone PSTN Internet End user SIP/PSTN gateway SipVxml SIP hardware phone Our Implementation of a SipVxml Browser. (Part of our CINEMA1 TestBed) Web server (HTTPd) [1] CINEMA - Columbia InterNet Extensible Multimedia Architecture SIP/VoiceXML @ Columbia University

  9. Conference server Conferencing 1. INVITE sipvxml 2. Call accepted 3. Enter your four digit PIN 4. Entered 4-6-8-3 5. Authenticate user, 4683=>Alice 6. Enter the conference identifier 7. Entered 2-3-# 8. Permission to join, 23=>meet SipVxml 9. REFER meet@conference 10.Terminate the old call Caller 11.INVITE meet@conference Call transfer vs bridged mode SIP/VoiceXML @ Columbia University

  10. Ease & Flexibility The ease & flexibility of SipVXML enables us to build custom telephonic applications to suit our needs. E.g Volume Check Application 1. INVITE sipvxml 2. Menu 1. Vol Check 2. Mic Check 3. User enters 2 4. User speaks out a voice sample 5. Voice sample is analyzed 6. SipVXML: Vol level too high/low/… 7. User adjusts the vol level. SipVxml 7. User now joins conference. Conference server Caller SIP/VoiceXML @ Columbia University

  11. More usage in the CINEMA test-bed • Unified messaging access • Email by phone • Event notification and scheduling • Audio volume level for conference • Advanced conference control SIP/VoiceXML @ Columbia University

  12. Conclusions • VoiceXML is simple and exciting • Sipvxml is useful for IP telephony and regular telephony • Numerous easy to develop applications http://www.cs.columbia.edu/IRT/cinema/doc/sipvxml.html http://www.cs.columbia.edu/IRT/cinema http://www.w3.org/Voice/ SIP/VoiceXML @ Columbia University

More Related